Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video calls seem to work well when both the devices are connected to the same network, however, when one device connects to a different network (the two devices now are on different networks), they are able to register on SIP server, and even call can be triggered and accepted between the two devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
Following is my kamailio.cfg file:
#!define WITH_DEBUG #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_NAT #!define WITH_USRLOCDB
tcp_async = yes tcp_connect_timeout=5
#!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: sr-users@lists.sip-router.org # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif
####### Include Local Config If Exists ######### import_file "kamailio-local.cfg"
####### Defined Values #########
# *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://root:root@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif
# - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5
#!define FLB_NATB 6 #!define FLB_NATSIPPING 7
####### Global Parameters #########
### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif
memdbg=5 memlog=5
log_facility=LOG_LOCAL0
fork=yes children=4
/* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes
/* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no
/* add local domain aliases */ alias="xyz.com"
/* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 listen=tcp:172.31.47.138:5060 advertise 54.191.193.***:5060
/* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060
#!ifdef WITH_TLS enable_tls=yes #!endif
# life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605
####### Custom Parameters #########
# These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id #
#!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif
#!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif
####### Modules Section ########
# set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/local/lib64/kamailio/modules/" #!endif
#!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif
loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so"
#!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif
#!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif
#!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif
#!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif
#!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif
#!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif
#!ifdef WITH_TLS loadmodule "tls.so" #!endif
#!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif
#!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif
#!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000)
# ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0)
# ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0)
# ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif
# ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif
# ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN)
# ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif
#!endif
# ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif
# ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif
# ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif
#!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL)
# ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif
#!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
# ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:xyz.com")
# params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif
#!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif
#!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4)
# ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif
#!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif
#!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif
####### Routing Logic ########
# Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route {
# per request initial checks route(REQINIT);
# NAT detection route(NATDETECT);
# CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; }
# handle requests within SIP dialogs route(WITHINDLG);
### only initial requests (no To tag)
t_check_trans();
# authentication route(AUTH);
# record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route();
# account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting }
# dispatch requests to foreign domains route(SIPOUT);
### requests for my local domains
# handle presence related requests route(PRESENCE);
# handle registrations route(REGISTRAR);
if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }
# dispatch destinations to PSTN route(PSTN);
# user location service route(LOCATION); }
route[RELAY] {
# enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); }
if (!t_relay()) { sl_reply_error(); } exit; }
# Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif
if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }
if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } }
# Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } }
# Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error();
exit; } }
# USER location service route[LOCATION] {
#!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif
#!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif
$avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } }
# when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); }
route(RELAY); exit; }
# Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return;
if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; }
#!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }
if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif
# if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; }
# Authentication route route[AUTH] { #!ifdef WITH_AUTH
#!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif
if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; }
#!endif return; }
# Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; }
# RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;
rtpproxy_manage("co");
if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; }
# URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; }
# Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } }
# PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; }
# route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(+|00)[1-9][0-9]{3,20}$")) return;
# only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; }
if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); }
route(RELAY); exit; #!endif
return; }
# XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif
# route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return;
# check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif
return; }
# manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); }
# manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); }
# manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE);
if (t_is_canceled()) { exit; }
#!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif
#!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }
Hello,
no time to look at config, but if you run the sip server on a private IP behind a port forwarding address, you have to use also rtpproxy with advertising address -- see the second parameter of rtpproxy_manage() or search on the web for a patch to rtpproxy to add advertising address via command line parameter.
Cheers, Daniel
On 03/09/14 12:23, Abhishek Saini wrote:
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video calls seem to work well when both the devices are connected to the same network, however, when one device connects to a different network (the two devices now are on different networks), they are able to register on SIP server, and even call can be triggered and accepted between the two devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
Following is my kamailio.cfg file:
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have described (advertising address - found that after posting the comment). But it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
Regards, Abhishek
On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
Hello,
no time to look at config, but if you run the sip server on a private IP behind a port forwarding address, you have to use also rtpproxy with advertising address -- see the second parameter of rtpproxy_manage() or search on the web for a patch to rtpproxy to add advertising address via command line parameter.
Cheers, Daniel
On 03/09/14 12:23, Abhishek Saini wrote:
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video calls seem to work well when both the devices are connected to the same network, however, when one device connects to a different network (the two devices now are on different networks), they are able to register on SIP server, and even call can be triggered and accepted between the two devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
Following is my kamailio.cfg file:
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
On 04/09/14 09:20, Abhishek Saini wrote:
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have described (advertising address - found that after posting the comment). But it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
use ngrep to look at sip traffic, like:
ngrep -d any -qt -W byline port 5060
If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in the SDP. Also, the media IP in SDP should change from incoming INVITE to what is sent out in the IP of rtpproxy.
Cheers, Daniel
Regards, Abhishek
On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com> wrote:
Hello, no time to look at config, but if you run the sip server on a private IP behind a port forwarding address, you have to use also rtpproxy with advertising address -- see the second parameter of rtpproxy_manage() or search on the web for a patch to rtpproxy to add advertising address via command line parameter. Cheers, Daniel On 03/09/14 12:23, Abhishek Saini wrote: Hi, I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video calls seem to work well when both the devices are connected to the same network, however, when one device connects to a different network (the two devices now are on different networks), they are able to register on SIP server, and even call can be triggered and accepted between the two devices but there is no video/audio transmission. I have setup rtpproxy but i don't know whether it's working or not. Any help on this would be highly appreciated. Following is my kamailio.cfg file: -- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Daniel,
Thanks, i was able to use the command you provided, but did not find the chunks you have specified(a=nortproxy:yes (iirc)) in the data. Checked by calling from webrtc client to a desktop client(blink).
When is rtpproxy used though? Kamailio says that it only transmits SIP signals and has not much to do with the media(voice or video). So, that means, it utilizes the rtpproxy to transmit the SIP signals(for non-symmetric NAT), If so then i think, the rtpproxy is working fine, as i have always been able to make and receive calls and only the media (voice or video) are not working (cross network).
I have also setup webrtc - it's working fine (firefox to firefox) but when i call from firefox to desktop client, it does not work(only rings, but does not connect). I read about webrtc_breaker but there does not seem to be a module for that in kamailio.
I think these two issues are somehow interlinked, please suggest me on this.
Regards, Abhishek
On Thu, Sep 4, 2014 at 1:28 PM, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
On 04/09/14 09:20, Abhishek Saini wrote:
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have described (advertising address - found that after posting the comment). But it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
use ngrep to look at sip traffic, like:
ngrep -d any -qt -W byline port 5060
If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in the SDP. Also, the media IP in SDP should change from incoming INVITE to what is sent out in the IP of rtpproxy.
Cheers, Daniel
Regards, Abhishek
On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
no time to look at config, but if you run the sip server on a private IP behind a port forwarding address, you have to use also rtpproxy with advertising address -- see the second parameter of rtpproxy_manage() or search on the web for a patch to rtpproxy to add advertising address via command line parameter.
Cheers, Daniel
On 03/09/14 12:23, Abhishek Saini wrote:
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video calls seem to work well when both the devices are connected to the same network, however, when one device connects to a different network (the two devices now are on different networks), they are able to register on SIP server, and even call can be triggered and accepted between the two devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
Following is my kamailio.cfg file:
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
Hello,
maybe you can send to mailing list the output of ngrep so we can look and check if a rtp relay is used.
If you need to bridge webrtc to classic sip phone, you have to use rtpengine.
Cheers, Daniel
On 04/09/14 13:01, Abhishek Saini wrote:
Hi Daniel,
Thanks, i was able to use the command you provided, but did not find the chunks you have specified(a=nortproxy:yes (iirc)) in the data. Checked by calling from webrtc client to a desktop client(blink).
When is rtpproxy used though? Kamailio says that it only transmits SIP signals and has not much to do with the media(voice or video). So, that means, it utilizes the rtpproxy to transmit the SIP signals(for non-symmetric NAT), If so then i think, the rtpproxy is working fine, as i have always been able to make and receive calls and only the media (voice or video) are not working (cross network).
I have also setup webrtc - it's working fine (firefox to firefox) but when i call from firefox to desktop client, it does not work(only rings, but does not connect). I read about webrtc_breaker but there does not seem to be a module for that in kamailio.
I think these two issues are somehow interlinked, please suggest me on this.
Regards, Abhishek
On Thu, Sep 4, 2014 at 1:28 PM, Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com> wrote:
Hello, On 04/09/14 09:20, Abhishek Saini wrote:
Hi Daniel, Thanks for reply. I did install patched rtpproxy and did configure it the way you have described (advertising address - found that after posting the comment). But it still does not seem to work. I don't quite know how can i debug, if rtpproxy is actually being used.
use ngrep to look at sip traffic, like: ngrep -d any -qt -W byline port 5060 If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in the SDP. Also, the media IP in SDP should change from incoming INVITE to what is sent out in the IP of rtpproxy. Cheers, Daniel
Regards, Abhishek On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla <miconda@gmail.com <mailto:miconda@gmail.com>> wrote: Hello, no time to look at config, but if you run the sip server on a private IP behind a port forwarding address, you have to use also rtpproxy with advertising address -- see the second parameter of rtpproxy_manage() or search on the web for a patch to rtpproxy to add advertising address via command line parameter. Cheers, Daniel On 03/09/14 12:23, Abhishek Saini wrote: Hi, I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video calls seem to work well when both the devices are connected to the same network, however, when one device connects to a different network (the two devices now are on different networks), they are able to register on SIP server, and even call can be triggered and accepted between the two devices but there is no video/audio transmission. I have setup rtpproxy but i don't know whether it's working or not. Any help on this would be highly appreciated. Following is my kamailio.cfg file: -- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany
Hi,
Please find attached the output of ngrep for three type of combinations/connections:
key: Blink is the desktop sip client and ntw means network.
blink2blink_same_ntw_successful webrtc2blink_same_ntw_failed webrtc2webrtc_same_ntw_successful
We also need to enable webrtc to classic sip phone calls, like on iphones/desktops etc. I could not find a good tutorial on rtpengine, and the steps to replace rtpproxy with rtpengine.
Please suggest me on this.
Regards, Abhishek
On Thu, Sep 4, 2014 at 6:02 PM, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
maybe you can send to mailing list the output of ngrep so we can look and check if a rtp relay is used.
If you need to bridge webrtc to classic sip phone, you have to use rtpengine.
Cheers, Daniel
On 04/09/14 13:01, Abhishek Saini wrote:
Hi Daniel,
Thanks, i was able to use the command you provided, but did not find the chunks you have specified(a=nortproxy:yes (iirc)) in the data. Checked by calling from webrtc client to a desktop client(blink).
When is rtpproxy used though? Kamailio says that it only transmits SIP signals and has not much to do with the media(voice or video). So, that means, it utilizes the rtpproxy to transmit the SIP signals(for non-symmetric NAT), If so then i think, the rtpproxy is working fine, as i have always been able to make and receive calls and only the media (voice or video) are not working (cross network).
I have also setup webrtc - it's working fine (firefox to firefox) but when i call from firefox to desktop client, it does not work(only rings, but does not connect). I read about webrtc_breaker but there does not seem to be a module for that in kamailio.
I think these two issues are somehow interlinked, please suggest me on this.
Regards, Abhishek
On Thu, Sep 4, 2014 at 1:28 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
On 04/09/14 09:20, Abhishek Saini wrote:
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have described (advertising address - found that after posting the comment). But it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
use ngrep to look at sip traffic, like:
ngrep -d any -qt -W byline port 5060
If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in the SDP. Also, the media IP in SDP should change from incoming INVITE to what is sent out in the IP of rtpproxy.
Cheers, Daniel
Regards, Abhishek
On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
no time to look at config, but if you run the sip server on a private IP behind a port forwarding address, you have to use also rtpproxy with advertising address -- see the second parameter of rtpproxy_manage() or search on the web for a patch to rtpproxy to add advertising address via command line parameter.
Cheers, Daniel
On 03/09/14 12:23, Abhishek Saini wrote:
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video calls seem to work well when both the devices are connected to the same network, however, when one device connects to a different network (the two devices now are on different networks), they are able to register on SIP server, and even call can be triggered and accepted between the two devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
Following is my kamailio.cfg file:
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
Hi,
I have not heard on my last reply (it went in moderation). So, I am posting one ngrep result here, Please let me know on this:
interface: any filter: (ip or ip6) and ( port 5060 ) # T 2014/09/04 12:51:26.423430 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] INVITE sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com. Contact: sip:48795260@192.168.0.3:1941;transport=tcp. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14955 INVITE. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Supported: 100rel, replaces, norefersub, gruu. User-Agent: Blink 0.9.1.2 (Windows). Content-Type: application/sdp. Content-Length: 639. . v=0. o=- 3618843694 3618843694 IN IP4 192.168.0.3. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101. c=IN IP4 192.168.0.3. a=rtcp:50037. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:9 G722/8000. a=rtpmap:104 speex/32000. a=rtpmap:103 speex/16000. a=rtpmap:3 GSM/8000. a=rtpmap:109 iLBC/8000. a=fmtp:109 mode=20. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT. a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7Ai8sovJF. a=sendrecv.
# T 2014/09/04 12:51:26.424351 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias;received=182.64.39.131. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.95b0. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14955 INVITE. Proxy-Authenticate: Digest realm="abc.com", nonce="VAhhelQIYE7RYMryoXu3/3LAv3hJc0hc". Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:26.761371 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] ACK sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.95b0. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14955 ACK. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:27.144744 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] INVITE sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com. Contact: sip:48795260@192.168.0.3:1941;transport=tcp. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 INVITE. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Supported: 100rel, replaces, norefersub, gruu. User-Agent: Blink 0.9.1.2 (Windows). Proxy-Authorization: Digest username="admin", realm="abc.com", nonce="VAhhelQIYE7RYMryoXu3/3LAv3hJc0hc", uri="sip:hari@abc.com", response="f3286a5acf46bf90ae6962ab9fe37f25". Content-Type: application/sdp. Content-Length: 639. . v=0. o=- 3618843694 3618843694 IN IP4 192.168.0.3. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101. c=IN IP4 192.168.0.3. a=rtcp:50037. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:9 G722/8000. a=rtpmap:104 speex/32000. a=rtpmap:103 speex/16000. a=rtpmap:3 GSM/8000. a=rtpmap:109 iLBC/8000. a=fmtp:109 mode=20. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT. a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7Ai8sovJF. a=sendrecv.
## T 2014/09/04 12:51:27.146564 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias;received=182.64.39.131. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 INVITE. Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:27.147226 172.31.47.138:5060 -> 182.64.39.131:59181 [A] INVITE sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Record-Route: sip:54.191.193.239;transport=tcp;lr=on;nat=yes. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com. Contact: sip:48795260@192.168.0.3:1941 ;transport=tcp;alias=182.64.39.131~3207~2. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 INVITE. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Supported: 100rel, replaces, norefersub, gruu. User-Agent: Blink 0.9.1.2 (Windows). Content-Type: application/sdp. Content-Length: 639. . v=0. o=- 3618843694 3618843694 IN IP4 192.168.0.3. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101. c=IN IP4 192.168.0.3. a=rtcp:50037. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:9 G722/8000. a=rtpmap:104 speex/32000. a=rtpmap:103 speex/16000. a=rtpmap:3 GSM/8000. a=rtpmap:109 iLBC/8000. a=fmtp:109 mode=20. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT. a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7A # T 2014/09/04 12:51:27.147238 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] i8sovJF. a=sendrecv.
# T 2014/09/04 12:51:27.465870 182.64.39.131:59181 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:27.475358 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP 54.191.193.239:5060 ;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:27.681136 182.64.39.131:3207 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:27.839852 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SIP/2.0 180 Ringing. Via: SIP/2.0/TCP 54.191.193.239:5060 ;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Contact: sip:80523769@192.168.0.6:59180;transport=tcp. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Content-Length: 0. .
## T 2014/09/04 12:51:27.840499 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 180 Ringing. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Contact: sip:80523769@192.168.0.6:59180 ;transport=tcp;alias=182.64.39.131~59181~2. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Content-Length: 0. .
# T 2014/09/04 12:51:28.373579 182.64.39.131:3207 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:30.264254 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] PUBLISH sip:admin@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj5084fdd1e5714d2d8f3182b9fbb8efc4;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=1b4d86568a1a4293bf7b04d0f0a4bb9f. To: "abhishek" sip:admin@abc.com. Call-ID: 100497fd86d14652a53347f3c5930ef3. CSeq: 1 PUBLISH. Event: presence. Expires: 600. User-Agent: Blink 0.9.1.2 (Windows). Content-Type: application/pidf+xml. Content-Length: 761. . <?xml version='1.0' encoding='UTF-8'?> <presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Aadmin%40abc.com"><tuple id="SID-8ee49af9e5034405300e43c6ecc590dd"><status><basic>closed</basic>agp-pidf:extendedoffline</agp-pidf:extended></status>caps:servcaps/<contact>sip%3Aadmin% 40abc.com</contact><timestamp>2014-09-04T16:17:08.257625+05:30</timestamp></tuple>dm:person id="PID-8ee49af9e5034405300e43c6ecc590dd"rpid:activitiesrpid:otheroffline</rpid:other></rpid:activities>dm:timestamp2014-09-04T16:17:08.257625+05:30</dm:timestamp></dm:person></presence> # T 2014/09/04 12:51:30.264906 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;branch=z9hG4bKPj5084fdd1e5714d2d8f3182b9fbb8efc4;alias;received=182.64.39.131. From: "abhishek" sip:admin@abc.com;tag=1b4d86568a1a4293bf7b04d0f0a4bb9f. To: "abhishek" <sip:admin@abc.com
;tag=16061544dcc5db830f3ff5cfaeeb9db0.4dd8.
Call-ID: 100497fd86d14652a53347f3c5930ef3. CSeq: 1 PUBLISH. Proxy-Authenticate: Digest realm="abc.com", nonce="VAhhflQIYFLstGA4jrcYrZLssnXzkltX". Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:30.611704 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] PUBLISH sip:admin@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj6a7a9b0d3bcc4d84af4ceaa7c09c1efb;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=1b4d86568a1a4293bf7b04d0f0a4bb9f. To: "abhishek" sip:admin@abc.com. Call-ID: 100497fd86d14652a53347f3c5930ef3. CSeq: 2 PUBLISH. Event: presence. Expires: 600. User-Agent: Blink 0.9.1.2 (Windows). Proxy-Authorization: Digest username="admin", realm="abc.com", nonce="VAhhflQIYFLstGA4jrcYrZLssnXzkltX", uri="sip:admin@abc.com", response="ac284e634f819e5db0b8502e4ee8e8bf". Content-Type: application/pidf+xml. Content-Length: 761. . <?xml version='1.0' encoding='UTF-8'?> <presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Aadmin%40abc.com"><tuple id="SID-8ee49af9e5034405300e43c6ecc590dd"><status><basic>closed</basic>agp-pidf:extendedoffline</agp-pidf:extended></status>caps:servcaps/<contact>sip%3Aadmin% 40abc.com</contact><timestamp>2014-09-04T16:17:08.257625+05:30</timestamp></tuple>dm:person id="PID-8ee49af9e5034405300e43c6ecc590dd"rpid:activitiesrpid:otheroffline</rpid:other></rpid:activities>dm:timestamp2014-09-04T16:17:08.257625+05:30</dm:timestamp></dm:person></presence> # T 2014/09/04 12:51:30.612812 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 404 Not here. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;branch=z9hG4bKPj6a7a9b0d3bcc4d84af4ceaa7c09c1efb;alias;received=182.64.39.131. From: "abhishek" sip:admin@abc.com;tag=1b4d86568a1a4293bf7b04d0f0a4bb9f. To: "abhishek" <sip:admin@abc.com
;tag=16061544dcc5db830f3ff5cfaeeb9db0.fc61.
Call-ID: 100497fd86d14652a53347f3c5930ef3. CSeq: 2 PUBLISH. Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:31.146313 182.64.39.131:3207 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:34.066168 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 54.191.193.239:5060 ;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Contact: sip:80523769@192.168.0.6:59180;transport=tcp. Supported: 100rel, replaces, norefersub, gruu. Content-Type: application/sdp. Content-Length: 352. . v=0. o=- 3618843693 3618843694 IN IP4 192.168.0.6. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50000 RTP/AVP 113 101. c=IN IP4 192.168.0.6. a=rtcp:50001. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1. a=sendrecv.
## T 2014/09/04 12:51:34.067096 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Contact: sip:80523769@192.168.0.6:59180 ;transport=tcp;alias=182.64.39.131~59181~2. Supported: 100rel, replaces, norefersub, gruu. Content-Type: application/sdp. Content-Length: 352. . v=0. o=- 3618843693 3618843694 IN IP4 192.168.0.6. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50000 RTP/AVP 113 101. c=IN IP4 192.168.0.6. a=rtcp:50001. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1. a=sendrecv.
# T 2014/09/04 12:51:34.408348 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp;alias=182.64.39.131~59181~2 SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:34.409438 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:34.573751 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 54.191.193.239:5060 ;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Contact: sip:80523769@192.168.0.6:59180;transport=tcp. Supported: 100rel, replaces, norefersub, gruu. Content-Type: application/sdp. Content-Length: 352. . v=0. o=- 3618843693 3618843694 IN IP4 192.168.0.6. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50000 RTP/AVP 113 101. c=IN IP4 192.168.0.6. a=rtcp:50001. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1. a=sendrecv.
## T 2014/09/04 12:51:34.574344 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Contact: sip:80523769@192.168.0.6:59180 ;transport=tcp;alias=182.64.39.131~59181~2. Supported: 100rel, replaces, norefersub, gruu. Content-Type: application/sdp. Content-Length: 352. . v=0. o=- 3618843693 3618843694 IN IP4 192.168.0.6. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50000 RTP/AVP 113 101. c=IN IP4 192.168.0.6. a=rtcp:50001. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1. a=sendrecv.
# T 2014/09/04 12:51:34.778979 182.64.39.131:59181 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:34.912960 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp;alias=182.64.39.131~59181~2 SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:34.913953 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:35.502496 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:36.094501 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
### T 2014/09/04 12:51:44.484915 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] BYE sip:80523769@192.168.0.6:59180;transport=tcp;alias=182.64.39.131~59181~2 SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14957 BYE. Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:44.486679 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] BYE sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKe247.6a1626e164da46d628f73b13d708c96b.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14957 BYE. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:44.815901 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 54.191.193.239:5060 ;received=54.191.193.239;branch=z9hG4bKe247.6a1626e164da46d628f73b13d708c96b.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14957 BYE. Server: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:44.816640 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14957 BYE. Server: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:45.352809 182.64.39.131:3207 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:47.241230 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] PUBLISH sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport;branch=z9hG4bKPjd05739db2b0d4e50a4917e5a0c86d4b8;alias. Max-Forwards: 70. From: "Hari" sip:hari@abc.com;tag=6ff49f144b63465ba09cf2152389be9d. To: "Hari" sip:hari@abc.com. Call-ID: 47794de446374174a07d801f1f2a0965. CSeq: 1 PUBLISH. Event: presence. Expires: 600. User-Agent: Blink 0.9.1.2 (Windows). Content-Type: application/pidf+xml. Content-Length: 759. . <?xml version='1.0' encoding='UTF-8'?> <presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Ahari%40abc.com"><tuple id="SID-423f2424f8cc3a5eacb0ebcb52eef7ea"><status><basic>closed</basic>agp-pidf:extendedoffline</agp-pidf:extended></status>caps:servcaps/<contact>sip%3Ahari% 40abc.com</contact><timestamp>2014-09-04T18:17:14.328500+05:30</timestamp></tuple>dm:person id="PID-423f2424f8cc3a5eacb0ebcb52eef7ea"rpid:activitiesrpid:otheroffline</rpid:other></rpid:activities>dm:timestamp2014-09-04T18:17:14.328500+05:30</dm:timestamp></dm:person></presence> ## T 2014/09/04 12:51:47.242107 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport=59181;branch=z9hG4bKPjd05739db2b0d4e50a4917e5a0c86d4b8;alias;received=182.64.39.131. From: "Hari" sip:hari@abc.com;tag=6ff49f144b63465ba09cf2152389be9d. To: "Hari" sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.682e. Call-ID: 47794de446374174a07d801f1f2a0965. CSeq: 1 PUBLISH. Proxy-Authenticate: Digest realm="abc.com", nonce="VAhhj1QIYGNmIT4Bvfge3Gurx+i9VnaY". Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:47.575172 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] PUBLISH sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport;branch=z9hG4bKPj51cb838a71654684bc99143ca42d929b;alias. Max-Forwards: 70. From: "Hari" sip:hari@abc.com;tag=6ff49f144b63465ba09cf2152389be9d. To: "Hari" sip:hari@abc.com. Call-ID: 47794de446374174a07d801f1f2a0965. CSeq: 2 PUBLISH. Event: presence. Expires: 600. User-Agent: Blink 0.9.1.2 (Windows). Proxy-Authorization: Digest username="hari", realm="abc.com", nonce="VAhhj1QIYGNmIT4Bvfge3Gurx+i9VnaY", uri="sip:hari@abc.com", response="30700144d91c2d6f88b35b3dc6272170". Content-Type: application/pidf+xml. Content-Length: 759. . <?xml version='1.0' encoding='UTF-8'?> <presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Ahari%40abc.com"><tuple id="SID-423f2424f8cc3a5eacb0ebcb52eef7ea"><status><basic>closed</basic>agp-pidf:extendedoffline</agp-pidf:extended></status>caps:servcaps/<contact>sip%3Ahari% 40abc.com</contact><timestamp>2014-09-04T18:17:14.328500+05:30</timestamp></tuple>dm:person id="PID-423f2424f8cc3a5eacb0ebcb52eef7ea"rpid:activitiesrpid:otheroffline</rpid:other></rpid:activities>dm:timestamp2014-09-04T18:17:14.328500+05:30</dm:timestamp></dm:person></presence> # T 2014/09/04 12:51:47.576515 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] SIP/2.0 404 Not here. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport=59181;branch=z9hG4bKPj51cb838a71654684bc99143ca42d929b;alias;received=182.64.39.131. From: "Hari" sip:hari@abc.com;tag=6ff49f144b63465ba09cf2152389be9d. To: "Hari" sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.29b9. Call-ID: 47794de446374174a07d801f1f2a0965. CSeq: 2 PUBLISH. Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:47.954792 182.64.39.131:59181 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:51.297577 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SUBSCRIBE sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport;branch=z9hG4bKPje26aae02a5aa421ea9b1dde05a016fd4;alias. Max-Forwards: 70. From: "Hari" sip:hari@abc.com;tag=bb028567e3c24d6792830754d4894f7d. To: sip:hari@abc.com. Contact: sip:80523769@192.168.0.6:59180;transport=tcp. Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389. CSeq: 14041 SUBSCRIBE. Event: message-summary. Expires: 600. Supported: 100rel, replaces, norefersub, gruu. Accept: application/simple-message-summary. Allow-Events: conference, message-summary, dialog, presence, presence.winfo, xcap-diff, dialog.winfo, refer. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:51.298522 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport=59181;branch=z9hG4bKPje26aae02a5aa421ea9b1dde05a016fd4;alias;received=182.64.39.131. From: "Hari" sip:hari@abc.com;tag=bb028567e3c24d6792830754d4894f7d. To: sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.95ac. Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389. CSeq: 14041 SUBSCRIBE. Proxy-Authenticate: Digest realm="abc.com", nonce="VAhhk1QIYGdyckAkr0vq+jaOzndpKg2N". Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:51.629211 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SUBSCRIBE sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport;branch=z9hG4bKPjb18a553f47ab4eec9f511d927a7bdd95;alias. Max-Forwards: 70. From: "Hari" sip:hari@abc.com;tag=bb028567e3c24d6792830754d4894f7d. To: sip:hari@abc.com. Contact: sip:80523769@192.168.0.6:59180;transport=tcp. Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389. CSeq: 14042 SUBSCRIBE. Event: message-summary. Expires: 600. Supported: 100rel, replaces, norefersub, gruu. Accept: application/simple-message-summary. Allow-Events: conference, message-summary, dialog, presence, presence.winfo, xcap-diff, dialog.winfo, refer. User-Agent: Blink 0.9.1.2 (Windows). Proxy-Authorization: Digest username="hari", realm="abc.com", nonce="VAhhk1QIYGdyckAkr0vq+jaOzndpKg2N", uri="sip:hari@abc.com", response="a231fc305242b93cbdcbf58d36c67376". Content-Length: 0. .
# T 2014/09/04 12:51:51.630383 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] SIP/2.0 404 No voicemail service. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport=59181;branch=z9hG4bKPjb18a553f47ab4eec9f511d927a7bdd95;alias;received=182.64.39.131. From: "Hari" sip:hari@abc.com;tag=bb028567e3c24d6792830754d4894f7d. To: sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.ddf9. Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389. CSeq: 14042 SUBSCRIBE. Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:52.001491 182.64.39.131:59181 -> 172.31.47.138:5060 [A] ......
Regards
On Thu, Sep 4, 2014 at 6:54 PM, Abhishek Saini < abhishek.saini@enukesoftware.com> wrote:
Hi,
Please find attached the output of ngrep for three type of combinations/connections:
key: Blink is the desktop sip client and ntw means network.
blink2blink_same_ntw_successful webrtc2blink_same_ntw_failed webrtc2webrtc_same_ntw_successful
We also need to enable webrtc to classic sip phone calls, like on iphones/desktops etc. I could not find a good tutorial on rtpengine, and the steps to replace rtpproxy with rtpengine.
Please suggest me on this.
Regards, Abhishek
On Thu, Sep 4, 2014 at 6:02 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
maybe you can send to mailing list the output of ngrep so we can look and check if a rtp relay is used.
If you need to bridge webrtc to classic sip phone, you have to use rtpengine.
Cheers, Daniel
On 04/09/14 13:01, Abhishek Saini wrote:
Hi Daniel,
Thanks, i was able to use the command you provided, but did not find the chunks you have specified(a=nortproxy:yes (iirc)) in the data. Checked by calling from webrtc client to a desktop client(blink).
When is rtpproxy used though? Kamailio says that it only transmits SIP signals and has not much to do with the media(voice or video). So, that means, it utilizes the rtpproxy to transmit the SIP signals(for non-symmetric NAT), If so then i think, the rtpproxy is working fine, as i have always been able to make and receive calls and only the media (voice or video) are not working (cross network).
I have also setup webrtc - it's working fine (firefox to firefox) but when i call from firefox to desktop client, it does not work(only rings, but does not connect). I read about webrtc_breaker but there does not seem to be a module for that in kamailio.
I think these two issues are somehow interlinked, please suggest me on this.
Regards, Abhishek
On Thu, Sep 4, 2014 at 1:28 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
On 04/09/14 09:20, Abhishek Saini wrote:
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have described (advertising address - found that after posting the comment). But it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
use ngrep to look at sip traffic, like:
ngrep -d any -qt -W byline port 5060
If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in the SDP. Also, the media IP in SDP should change from incoming INVITE to what is sent out in the IP of rtpproxy.
Cheers, Daniel
Regards, Abhishek
On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
no time to look at config, but if you run the sip server on a private IP behind a port forwarding address, you have to use also rtpproxy with advertising address -- see the second parameter of rtpproxy_manage() or search on the web for a patch to rtpproxy to add advertising address via command line parameter.
Cheers, Daniel
On 03/09/14 12:23, Abhishek Saini wrote:
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video calls seem to work well when both the devices are connected to the same network, however, when one device connects to a different network (the two devices now are on different networks), they are able to register on SIP server, and even call can be triggered and accepted between the two devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
Following is my kamailio.cfg file:
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
Hi,
It appears that my last two messages have gone in moderation. Anyways, Can you please tell me, how can i setup rtpengine on Ubuntu machine? After installation - What configurations will i have to change?
I have lurked the internet a lot but did not find any tutorial on this. Would appreciate any help on this.
Regards
On Mon, Sep 8, 2014 at 10:14 AM, Abhishek Saini < abhishek.saini@enukesoftware.com> wrote:
Hi,
I have not heard on my last reply (it went in moderation). So, I am posting one ngrep result here, Please let me know on this:
interface: any filter: (ip or ip6) and ( port 5060 ) # T 2014/09/04 12:51:26.423430 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] INVITE sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com. Contact: sip:48795260@192.168.0.3:1941;transport=tcp. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14955 INVITE. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Supported: 100rel, replaces, norefersub, gruu. User-Agent: Blink 0.9.1.2 (Windows). Content-Type: application/sdp. Content-Length: 639. . v=0. o=- 3618843694 3618843694 IN IP4 192.168.0.3. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101. c=IN IP4 192.168.0.3. a=rtcp:50037. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:9 G722/8000. a=rtpmap:104 speex/32000. a=rtpmap:103 speex/16000. a=rtpmap:3 GSM/8000. a=rtpmap:109 iLBC/8000. a=fmtp:109 mode=20. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT. a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7Ai8sovJF. a=sendrecv.
# T 2014/09/04 12:51:26.424351 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias;received=182.64.39.131. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.95b0. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14955 INVITE. Proxy-Authenticate: Digest realm="abc.com", nonce="VAhhelQIYE7RYMryoXu3/3LAv3hJc0hc". Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:26.761371 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] ACK sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj26fb26ab22104aac842ae38bd3fea246;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.95b0. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14955 ACK. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:27.144744 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] INVITE sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com. Contact: sip:48795260@192.168.0.3:1941;transport=tcp. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 INVITE. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Supported: 100rel, replaces, norefersub, gruu. User-Agent: Blink 0.9.1.2 (Windows). Proxy-Authorization: Digest username="admin", realm="abc.com", nonce="VAhhelQIYE7RYMryoXu3/3LAv3hJc0hc", uri="sip:hari@abc.com", response="f3286a5acf46bf90ae6962ab9fe37f25". Content-Type: application/sdp. Content-Length: 639. . v=0. o=- 3618843694 3618843694 IN IP4 192.168.0.3. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101. c=IN IP4 192.168.0.3. a=rtcp:50037. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:9 G722/8000. a=rtpmap:104 speex/32000. a=rtpmap:103 speex/16000. a=rtpmap:3 GSM/8000. a=rtpmap:109 iLBC/8000. a=fmtp:109 mode=20. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT. a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7Ai8sovJF. a=sendrecv.
## T 2014/09/04 12:51:27.146564 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias;received=182.64.39.131. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 INVITE. Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:27.147226 172.31.47.138:5060 -> 182.64.39.131:59181 [A] INVITE sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Record-Route: sip:54.191.193.239;transport=tcp;lr=on;nat=yes. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com. Contact: sip:48795260@192.168.0.3:1941 ;transport=tcp;alias=182.64.39.131~3207~2. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 INVITE. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Supported: 100rel, replaces, norefersub, gruu. User-Agent: Blink 0.9.1.2 (Windows). Content-Type: application/sdp. Content-Length: 639. . v=0. o=- 3618843694 3618843694 IN IP4 192.168.0.3. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50036 RTP/AVP 113 9 104 103 3 109 0 8 101. c=IN IP4 192.168.0.3. a=rtcp:50037. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:9 G722/8000. a=rtpmap:104 speex/32000. a=rtpmap:103 speex/16000. a=rtpmap:3 GSM/8000. a=rtpmap:109 iLBC/8000. a=fmtp:109 mode=20. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fpeIqd7S4YbsjlTI+T3r/LBXQ+pAXB8j6upIi/zT. a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0lfQvO4gCr+Mx/177aH0128j8ghvylp7A # T 2014/09/04 12:51:27.147238 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] i8sovJF. a=sendrecv.
# T 2014/09/04 12:51:27.465870 182.64.39.131:59181 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:27.475358 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP 54.191.193.239:5060 ;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:27.681136 182.64.39.131:3207 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:27.839852 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SIP/2.0 180 Ringing. Via: SIP/2.0/TCP 54.191.193.239:5060 ;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Contact: sip:80523769@192.168.0.6:59180;transport=tcp. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Content-Length: 0. .
## T 2014/09/04 12:51:27.840499 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 180 Ringing. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Contact: sip:80523769@192.168.0.6:59180 ;transport=tcp;alias=182.64.39.131~59181~2. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Content-Length: 0. .
# T 2014/09/04 12:51:28.373579 182.64.39.131:3207 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:30.264254 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] PUBLISH sip:admin@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj5084fdd1e5714d2d8f3182b9fbb8efc4;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=1b4d86568a1a4293bf7b04d0f0a4bb9f. To: "abhishek" sip:admin@abc.com. Call-ID: 100497fd86d14652a53347f3c5930ef3. CSeq: 1 PUBLISH. Event: presence. Expires: 600. User-Agent: Blink 0.9.1.2 (Windows). Content-Type: application/pidf+xml. Content-Length: 761. .
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Aadmin%40abc.com"><tuple id="SID-8ee49af9e5034405300e43c6ecc590dd"><status><basic>closed</basic>agp-pidf:extendedoffline</agp-pidf:extended></status>caps:servcaps/<contact>sip%3Aadmin% 40abc.com</contact><timestamp>2014-09-04T16:17:08.257625+05:30</timestamp></tuple>dm:person id="PID-8ee49af9e5034405300e43c6ecc590dd"rpid:activitiesrpid:otheroffline</rpid:other></rpid:activities>dm:timestamp2014-09-04T16:17:08.257625+05:30</dm:timestamp></dm:person></presence> # T 2014/09/04 12:51:30.264906 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;branch=z9hG4bKPj5084fdd1e5714d2d8f3182b9fbb8efc4;alias;received=182.64.39.131. From: "abhishek" sip:admin@abc.com;tag=1b4d86568a1a4293bf7b04d0f0a4bb9f. To: "abhishek" <sip:admin@abc.com
;tag=16061544dcc5db830f3ff5cfaeeb9db0.4dd8.
Call-ID: 100497fd86d14652a53347f3c5930ef3. CSeq: 1 PUBLISH. Proxy-Authenticate: Digest realm="abc.com", nonce="VAhhflQIYFLstGA4jrcYrZLssnXzkltX". Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:30.611704 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] PUBLISH sip:admin@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj6a7a9b0d3bcc4d84af4ceaa7c09c1efb;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=1b4d86568a1a4293bf7b04d0f0a4bb9f. To: "abhishek" sip:admin@abc.com. Call-ID: 100497fd86d14652a53347f3c5930ef3. CSeq: 2 PUBLISH. Event: presence. Expires: 600. User-Agent: Blink 0.9.1.2 (Windows). Proxy-Authorization: Digest username="admin", realm="abc.com", nonce="VAhhflQIYFLstGA4jrcYrZLssnXzkltX", uri="sip:admin@abc.com", response="ac284e634f819e5db0b8502e4ee8e8bf". Content-Type: application/pidf+xml. Content-Length: 761. .
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Aadmin%40abc.com"><tuple id="SID-8ee49af9e5034405300e43c6ecc590dd"><status><basic>closed</basic>agp-pidf:extendedoffline</agp-pidf:extended></status>caps:servcaps/<contact>sip%3Aadmin% 40abc.com</contact><timestamp>2014-09-04T16:17:08.257625+05:30</timestamp></tuple>dm:person id="PID-8ee49af9e5034405300e43c6ecc590dd"rpid:activitiesrpid:otheroffline</rpid:other></rpid:activities>dm:timestamp2014-09-04T16:17:08.257625+05:30</dm:timestamp></dm:person></presence> # T 2014/09/04 12:51:30.612812 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 404 Not here. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;branch=z9hG4bKPj6a7a9b0d3bcc4d84af4ceaa7c09c1efb;alias;received=182.64.39.131. From: "abhishek" sip:admin@abc.com;tag=1b4d86568a1a4293bf7b04d0f0a4bb9f. To: "abhishek" <sip:admin@abc.com
;tag=16061544dcc5db830f3ff5cfaeeb9db0.fc61.
Call-ID: 100497fd86d14652a53347f3c5930ef3. CSeq: 2 PUBLISH. Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:31.146313 182.64.39.131:3207 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:34.066168 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 54.191.193.239:5060 ;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Contact: sip:80523769@192.168.0.6:59180;transport=tcp. Supported: 100rel, replaces, norefersub, gruu. Content-Type: application/sdp. Content-Length: 352. . v=0. o=- 3618843693 3618843694 IN IP4 192.168.0.6. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50000 RTP/AVP 113 101. c=IN IP4 192.168.0.6. a=rtcp:50001. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1. a=sendrecv.
## T 2014/09/04 12:51:34.067096 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Contact: sip:80523769@192.168.0.6:59180 ;transport=tcp;alias=182.64.39.131~59181~2. Supported: 100rel, replaces, norefersub, gruu. Content-Type: application/sdp. Content-Length: 352. . v=0. o=- 3618843693 3618843694 IN IP4 192.168.0.6. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50000 RTP/AVP 113 101. c=IN IP4 192.168.0.6. a=rtcp:50001. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1. a=sendrecv.
# T 2014/09/04 12:51:34.408348 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp;alias=182.64.39.131~59181~2 SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:34.409438 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:34.573751 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 54.191.193.239:5060 ;received=54.191.193.239;branch=z9hG4bKd247.56bfe8f59f4bfc2c5f783d4adb3e72e1.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Contact: sip:80523769@192.168.0.6:59180;transport=tcp. Supported: 100rel, replaces, norefersub, gruu. Content-Type: application/sdp. Content-Length: 352. . v=0. o=- 3618843693 3618843694 IN IP4 192.168.0.6. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50000 RTP/AVP 113 101. c=IN IP4 192.168.0.6. a=rtcp:50001. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1. a=sendrecv.
## T 2014/09/04 12:51:34.574344 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj4a167a7a11654ba7b08ac6dcefa85845;alias. Record-Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14956 INVITE. Server: Blink 0.9.1.2 (Windows). Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. Contact: sip:80523769@192.168.0.6:59180 ;transport=tcp;alias=182.64.39.131~59181~2. Supported: 100rel, replaces, norefersub, gruu. Content-Type: application/sdp. Content-Length: 352. . v=0. o=- 3618843693 3618843694 IN IP4 192.168.0.6. s=Blink 0.9.1.2 (Windows). t=0 0. m=audio 50000 RTP/AVP 113 101. c=IN IP4 192.168.0.6. a=rtcp:50001. a=rtpmap:113 opus/48000. a=fmtp:113 useinbandfec=1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y2a4eThKFFtUJ8pdGGQMr/MB0PDTDP0/fFkvK+H1. a=sendrecv.
# T 2014/09/04 12:51:34.778979 182.64.39.131:59181 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:34.912960 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp;alias=182.64.39.131~59181~2 SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:34.913953 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:35.502496 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:36.094501 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] ACK sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKd247.6fc7f29a9c75230c664d658814fcd105.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj093d16f04a8a4220b8dfc62cc00897cc;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14956 ACK. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
### T 2014/09/04 12:51:44.484915 182.64.39.131:3207 -> 172.31.47.138:5060 [AP] BYE sip:80523769@192.168.0.6:59180;transport=tcp;alias=182.64.39.131~59181~2 SIP/2.0. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias. Max-Forwards: 70. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14957 BYE. Route: sip:54.191.193.239;transport=tcp;lr;nat=yes. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:44.486679 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] BYE sip:80523769@192.168.0.6:59180;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 54.191.193.239:5060 ;branch=z9hG4bKe247.6a1626e164da46d628f73b13d708c96b.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;received=182.64.39.131;rport=3207;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias. Max-Forwards: 69. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. Call-ID: b0cac02326d44105aa008f9edd352e8e. CSeq: 14957 BYE. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:44.815901 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 54.191.193.239:5060 ;received=54.191.193.239;branch=z9hG4bKe247.6a1626e164da46d628f73b13d708c96b.0;i=f5. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14957 BYE. Server: Blink 0.9.1.2 (Windows). Content-Length: 0. .
## T 2014/09/04 12:51:44.816640 172.31.47.138:5060 -> 182.64.39.131:3207 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 192.168.0.3:3207 ;rport=3207;received=182.64.39.131;branch=z9hG4bKPj7f11c7ed0bae49b285640607420e8251;alias. Call-ID: b0cac02326d44105aa008f9edd352e8e. From: "abhishek" sip:admin@abc.com;tag=93d6b660540f40338f9ed66f946ae11a. To: sip:hari@abc.com;tag=94d1930ec5b84d5bbd1241ce9feab364. CSeq: 14957 BYE. Server: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:45.352809 182.64.39.131:3207 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:47.241230 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] PUBLISH sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport;branch=z9hG4bKPjd05739db2b0d4e50a4917e5a0c86d4b8;alias. Max-Forwards: 70. From: "Hari" sip:hari@abc.com;tag=6ff49f144b63465ba09cf2152389be9d. To: "Hari" sip:hari@abc.com. Call-ID: 47794de446374174a07d801f1f2a0965. CSeq: 1 PUBLISH. Event: presence. Expires: 600. User-Agent: Blink 0.9.1.2 (Windows). Content-Type: application/pidf+xml. Content-Length: 759. .
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Ahari%40abc.com"><tuple id="SID-423f2424f8cc3a5eacb0ebcb52eef7ea"><status><basic>closed</basic>agp-pidf:extendedoffline</agp-pidf:extended></status>caps:servcaps/<contact>sip%3Ahari% 40abc.com</contact><timestamp>2014-09-04T18:17:14.328500+05:30</timestamp></tuple>dm:person id="PID-423f2424f8cc3a5eacb0ebcb52eef7ea"rpid:activitiesrpid:otheroffline</rpid:other></rpid:activities>dm:timestamp2014-09-04T18:17:14.328500+05:30</dm:timestamp></dm:person></presence> ## T 2014/09/04 12:51:47.242107 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport=59181;branch=z9hG4bKPjd05739db2b0d4e50a4917e5a0c86d4b8;alias;received=182.64.39.131. From: "Hari" sip:hari@abc.com;tag=6ff49f144b63465ba09cf2152389be9d. To: "Hari" sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.682e. Call-ID: 47794de446374174a07d801f1f2a0965. CSeq: 1 PUBLISH. Proxy-Authenticate: Digest realm="abc.com", nonce="VAhhj1QIYGNmIT4Bvfge3Gurx+i9VnaY". Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:47.575172 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] PUBLISH sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport;branch=z9hG4bKPj51cb838a71654684bc99143ca42d929b;alias. Max-Forwards: 70. From: "Hari" sip:hari@abc.com;tag=6ff49f144b63465ba09cf2152389be9d. To: "Hari" sip:hari@abc.com. Call-ID: 47794de446374174a07d801f1f2a0965. CSeq: 2 PUBLISH. Event: presence. Expires: 600. User-Agent: Blink 0.9.1.2 (Windows). Proxy-Authorization: Digest username="hari", realm="abc.com", nonce="VAhhj1QIYGNmIT4Bvfge3Gurx+i9VnaY", uri="sip:hari@abc.com", response="30700144d91c2d6f88b35b3dc6272170". Content-Type: application/pidf+xml. Content-Length: 759. .
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3Ahari%40abc.com"><tuple id="SID-423f2424f8cc3a5eacb0ebcb52eef7ea"><status><basic>closed</basic>agp-pidf:extendedoffline</agp-pidf:extended></status>caps:servcaps/<contact>sip%3Ahari% 40abc.com</contact><timestamp>2014-09-04T18:17:14.328500+05:30</timestamp></tuple>dm:person id="PID-423f2424f8cc3a5eacb0ebcb52eef7ea"rpid:activitiesrpid:otheroffline</rpid:other></rpid:activities>dm:timestamp2014-09-04T18:17:14.328500+05:30</dm:timestamp></dm:person></presence> # T 2014/09/04 12:51:47.576515 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] SIP/2.0 404 Not here. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport=59181;branch=z9hG4bKPj51cb838a71654684bc99143ca42d929b;alias;received=182.64.39.131. From: "Hari" sip:hari@abc.com;tag=6ff49f144b63465ba09cf2152389be9d. To: "Hari" sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.29b9. Call-ID: 47794de446374174a07d801f1f2a0965. CSeq: 2 PUBLISH. Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:47.954792 182.64.39.131:59181 -> 172.31.47.138:5060 [A] ...... # T 2014/09/04 12:51:51.297577 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SUBSCRIBE sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport;branch=z9hG4bKPje26aae02a5aa421ea9b1dde05a016fd4;alias. Max-Forwards: 70. From: "Hari" sip:hari@abc.com;tag=bb028567e3c24d6792830754d4894f7d. To: sip:hari@abc.com. Contact: sip:80523769@192.168.0.6:59180;transport=tcp. Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389. CSeq: 14041 SUBSCRIBE. Event: message-summary. Expires: 600. Supported: 100rel, replaces, norefersub, gruu. Accept: application/simple-message-summary. Allow-Events: conference, message-summary, dialog, presence, presence.winfo, xcap-diff, dialog.winfo, refer. User-Agent: Blink 0.9.1.2 (Windows). Content-Length: 0. .
# T 2014/09/04 12:51:51.298522 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport=59181;branch=z9hG4bKPje26aae02a5aa421ea9b1dde05a016fd4;alias;received=182.64.39.131. From: "Hari" sip:hari@abc.com;tag=bb028567e3c24d6792830754d4894f7d. To: sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.95ac. Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389. CSeq: 14041 SUBSCRIBE. Proxy-Authenticate: Digest realm="abc.com", nonce="VAhhk1QIYGdyckAkr0vq+jaOzndpKg2N". Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:51.629211 182.64.39.131:59181 -> 172.31.47.138:5060 [AP] SUBSCRIBE sip:hari@abc.com SIP/2.0. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport;branch=z9hG4bKPjb18a553f47ab4eec9f511d927a7bdd95;alias. Max-Forwards: 70. From: "Hari" sip:hari@abc.com;tag=bb028567e3c24d6792830754d4894f7d. To: sip:hari@abc.com. Contact: sip:80523769@192.168.0.6:59180;transport=tcp. Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389. CSeq: 14042 SUBSCRIBE. Event: message-summary. Expires: 600. Supported: 100rel, replaces, norefersub, gruu. Accept: application/simple-message-summary. Allow-Events: conference, message-summary, dialog, presence, presence.winfo, xcap-diff, dialog.winfo, refer. User-Agent: Blink 0.9.1.2 (Windows). Proxy-Authorization: Digest username="hari", realm="abc.com", nonce="VAhhk1QIYGdyckAkr0vq+jaOzndpKg2N", uri="sip:hari@abc.com", response="a231fc305242b93cbdcbf58d36c67376". Content-Length: 0. .
# T 2014/09/04 12:51:51.630383 172.31.47.138:5060 -> 182.64.39.131:59181 [AP] SIP/2.0 404 No voicemail service. Via: SIP/2.0/TCP 192.168.0.6:59181 ;rport=59181;branch=z9hG4bKPjb18a553f47ab4eec9f511d927a7bdd95;alias;received=182.64.39.131. From: "Hari" sip:hari@abc.com;tag=bb028567e3c24d6792830754d4894f7d. To: sip:hari@abc.com;tag=16061544dcc5db830f3ff5cfaeeb9db0.ddf9. Call-ID: 8cb97ab4f8ba4332b0ca5dbc5d88f389. CSeq: 14042 SUBSCRIBE. Server: kamailio (4.1.5 (x86_64/linux)). Content-Length: 0. .
# T 2014/09/04 12:51:52.001491 182.64.39.131:59181 -> 172.31.47.138:5060 [A] ......
Regards
On Thu, Sep 4, 2014 at 6:54 PM, Abhishek Saini < abhishek.saini@enukesoftware.com> wrote:
Hi,
Please find attached the output of ngrep for three type of combinations/connections:
key: Blink is the desktop sip client and ntw means network.
blink2blink_same_ntw_successful webrtc2blink_same_ntw_failed webrtc2webrtc_same_ntw_successful
We also need to enable webrtc to classic sip phone calls, like on iphones/desktops etc. I could not find a good tutorial on rtpengine, and the steps to replace rtpproxy with rtpengine.
Please suggest me on this.
Regards, Abhishek
On Thu, Sep 4, 2014 at 6:02 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
maybe you can send to mailing list the output of ngrep so we can look and check if a rtp relay is used.
If you need to bridge webrtc to classic sip phone, you have to use rtpengine.
Cheers, Daniel
On 04/09/14 13:01, Abhishek Saini wrote:
Hi Daniel,
Thanks, i was able to use the command you provided, but did not find the chunks you have specified(a=nortproxy:yes (iirc)) in the data. Checked by calling from webrtc client to a desktop client(blink).
When is rtpproxy used though? Kamailio says that it only transmits SIP signals and has not much to do with the media(voice or video). So, that means, it utilizes the rtpproxy to transmit the SIP signals(for non-symmetric NAT), If so then i think, the rtpproxy is working fine, as i have always been able to make and receive calls and only the media (voice or video) are not working (cross network).
I have also setup webrtc - it's working fine (firefox to firefox) but when i call from firefox to desktop client, it does not work(only rings, but does not connect). I read about webrtc_breaker but there does not seem to be a module for that in kamailio.
I think these two issues are somehow interlinked, please suggest me on this.
Regards, Abhishek
On Thu, Sep 4, 2014 at 1:28 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
On 04/09/14 09:20, Abhishek Saini wrote:
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have described (advertising address - found that after posting the comment). But it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
use ngrep to look at sip traffic, like:
ngrep -d any -qt -W byline port 5060
If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in the SDP. Also, the media IP in SDP should change from incoming INVITE to what is sent out in the IP of rtpproxy.
Cheers, Daniel
Regards, Abhishek
On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
no time to look at config, but if you run the sip server on a private IP behind a port forwarding address, you have to use also rtpproxy with advertising address -- see the second parameter of rtpproxy_manage() or search on the web for a patch to rtpproxy to add advertising address via command line parameter.
Cheers, Daniel
On 03/09/14 12:23, Abhishek Saini wrote:
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video calls seem to work well when both the devices are connected to the same network, however, when one device connects to a different network (the two devices now are on different networks), they are able to register on SIP server, and even call can be triggered and accepted between the two devices but there is no video/audio transmission.
I have setup rtpproxy but i don't know whether it's working or not.
Any help on this would be highly appreciated.
Following is my kamailio.cfg file:
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
Hi,
I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng package on my ubuntu box. As suggested here: http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module, but still not able to connect the webrtc calls to classic sip phones (and vice-versa). Below is the sip message that is traced:
SIP/2.0 488 Not acceptable here. Via: SIP/2.0/TCP 54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$ Via: SIP/2.0/WS df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$ From: "admin" sip:admin@abc.com;tag=bzhwwG8nT2gFwwJgIyrz. To: sip:hari@abc.com;tag=OIllTQf. Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a. CSeq: 65463 INVITE. User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2). Supported: replaces, outbound. Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards, Abhishek
Hello,
the reply code indicates that the media type is not supported, thus there has been no gatewaying between webrtc and classic rtp. Just replacing rtpproxy with rtpengine is not enough, there are different parameters that have to be provided.
Searching on web, I see that Carlos has published a config for it, see: - https://github.com/caruizdiaz/kamailio-ws
Cheers, Daniel
On 15/09/14 12:58, Abhishek Saini wrote:
Hi,
I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng package on my ubuntu box. As suggested here: http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module, but still not able to connect the webrtc calls to classic sip phones (and vice-versa). Below is the sip message that is traced:
SIP/2.0 488 Not acceptable here. Via: SIP/2.0/TCP 54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$ Via: SIP/2.0/WS df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$ From: "admin" <sip:admin@abc.com mailto:sip%3Aadmin@abc.com>;tag=bzhwwG8nT2gFwwJgIyrz. To: <sip:hari@abc.com mailto:sip%3Ahari@abc.com>;tag=OIllTQf. Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a. CSeq: 65463 INVITE. User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2). Supported: replaces, outbound. Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards, Abhishek
Hi Daniel,
Thanks for this.
I took the entire config files and configured it as per my ips and ports, after doing that, still no call establishment(webrtc to classic sip phones and vice-versa). Following is what i get in kamailio.log:
rtpp_test(): rtp proxy udp:127.0.0.1:7722 found, support for it enabled ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' ERROR: <script>: ==> duri=[sip:nudg.com:5060 ;lr;sipml5-outbound;transport=tcp] INFO: <script>: Request coming from WS ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' INFO: <script>: Reply from softphone: 100
And this SIP message: SIP/2.0 603 Failed to get local SDP.
Regards, Abhishek
On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
Hello,
the reply code indicates that the media type is not supported, thus there has been no gatewaying between webrtc and classic rtp. Just replacing rtpproxy with rtpengine is not enough, there are different parameters that have to be provided.
Searching on web, I see that Carlos has published a config for it, see:
Cheers, Daniel
On 15/09/14 12:58, Abhishek Saini wrote:
Hi,
I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng package on my ubuntu box. As suggested here: http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module, but still not able to connect the webrtc calls to classic sip phones (and vice-versa). Below is the sip message that is traced:
SIP/2.0 488 Not acceptable here. Via: SIP/2.0/TCP 54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$ Via: SIP/2.0/WS df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$ From: "admin" sip:admin@abc.com;tag=bzhwwG8nT2gFwwJgIyrz. To: sip:hari@abc.com;tag=OIllTQf. Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a. CSeq: 65463 INVITE. User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2). Supported: replaces, outbound. Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards, Abhishek
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
Hi Daniel,
I was able to solve a fraction of my problem, Actually, the github link had used rtpengine.so and i was using rptproxy-ng.so, there is a difference in the flag conventions between the two; i modified that to achieve a little progress.
Now, i am able to call on webrtc(firefox) from sip phone. However, after accepting call, there is no audio, and disconnecting the call from either end does not disconnect the call.
When i try to call from webrtc(firefox) to sip phone, there is no signalling at all, and the sip phone to webrtc calls can't connect after that. (I analyzed that mediaproxy-ng/rtpengine process terminates and has to be started again)
Following are the links to my latest kamailio.cfg file and port trace log of sip messages. http://jmp.sh/o0apKgP http://jmp.sh/HXnFRQj
I am clueless at the moment!
Regards, Abhishek
On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini < abhishek.saini@enukesoftware.com> wrote:
Hi Daniel,
Thanks for this.
I took the entire config files and configured it as per my ips and ports, after doing that, still no call establishment(webrtc to classic sip phones and vice-versa). Following is what i get in kamailio.log:
rtpp_test(): rtp proxy udp:127.0.0.1:7722 found, support for it enabled ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' ERROR: <script>: ==> duri=[sip:nudg.com:5060 ;lr;sipml5-outbound;transport=tcp] INFO: <script>: Request coming from WS ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' INFO: <script>: Reply from softphone: 100
And this SIP message: SIP/2.0 603 Failed to get local SDP.
Regards, Abhishek
On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
the reply code indicates that the media type is not supported, thus there has been no gatewaying between webrtc and classic rtp. Just replacing rtpproxy with rtpengine is not enough, there are different parameters that have to be provided.
Searching on web, I see that Carlos has published a config for it, see:
Cheers, Daniel
On 15/09/14 12:58, Abhishek Saini wrote:
Hi,
I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng package on my ubuntu box. As suggested here: http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module, but still not able to connect the webrtc calls to classic sip phones (and vice-versa). Below is the sip message that is traced:
SIP/2.0 488 Not acceptable here. Via: SIP/2.0/TCP 54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$ Via: SIP/2.0/WS df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$ From: "admin" sip:admin@abc.com;tag=bzhwwG8nT2gFwwJgIyrz. To: sip:hari@abc.com;tag=OIllTQf. Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a. CSeq: 65463 INVITE. User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2). Supported: replaces, outbound. Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards, Abhishek
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
Hello,
maybe you should play with kamailio master branch (which is in testing phase before becoming 4.2) -- there you have the rtpengine -- and see if you get it working. Once that, you can look at using an older version, knowing you have it working and be able to compare. As I needed latest features, whenever I needed webrtc gatewaying, I used devel branch of rtpengine module.
Cheers, Daniel
On 16/09/14 14:24, Abhishek Saini wrote:
Hi Daniel,
I was able to solve a fraction of my problem, Actually, the github link had used rtpengine.so and i was using rptproxy-ng.so, there is a difference in the flag conventions between the two; i modified that to achieve a little progress.
Now, i am able to call on webrtc(firefox) from sip phone. However, after accepting call, there is no audio, and disconnecting the call from either end does not disconnect the call.
When i try to call from webrtc(firefox) to sip phone, there is no signalling at all, and the sip phone to webrtc calls can't connect after that. (I analyzed that mediaproxy-ng/rtpengine process terminates and has to be started again)
Following are the links to my latest kamailio.cfg file and port trace log of sip messages. http://jmp.sh/o0apKgP http://jmp.sh/HXnFRQj
I am clueless at the moment!
Regards, Abhishek
On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini <abhishek.saini@enukesoftware.com mailto:abhishek.saini@enukesoftware.com> wrote:
Hi Daniel, Thanks for this. I took the entire config files and configured it as per my ips and ports, after doing that, still no call establishment(webrtc to classic sip phones and vice-versa). Following is what i get in kamailio.log: rtpp_test(): rtp proxy <udp:127.0.0.1:7722 <http://127.0.0.1:7722>> found, support for it enabled ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' ERROR: <script>: ==> duri=[sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp] INFO: <script>: Request coming from WS ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' INFO: <script>: Reply from softphone: 100 And this SIP message: SIP/2.0 603 Failed to get local SDP. Regards, Abhishek On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla <miconda@gmail.com <mailto:miconda@gmail.com>> wrote: Hello, the reply code indicates that the media type is not supported, thus there has been no gatewaying between webrtc and classic rtp. Just replacing rtpproxy with rtpengine is not enough, there are different parameters that have to be provided. Searching on web, I see that Carlos has published a config for it, see: - https://github.com/caruizdiaz/kamailio-ws Cheers, Daniel On 15/09/14 12:58, Abhishek Saini wrote:
Hi, I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng package on my ubuntu box. As suggested here: http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html I have kept rtpproxy-ng's configuration same as the rtpproxy module, but still not able to connect the webrtc calls to classic sip phones (and vice-versa). Below is the sip message that is traced: SIP/2.0 488 Not acceptable here. Via: SIP/2.0/TCP 54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$ Via: SIP/2.0/WS df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$ From: "admin" <sip:admin@abc.com <mailto:sip%3Aadmin@abc.com>>;tag=bzhwwG8nT2gFwwJgIyrz. To: <sip:hari@abc.com <mailto:sip%3Ahari@abc.com>>;tag=OIllTQf. Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a. CSeq: 65463 INVITE. User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2). Supported: replaces, outbound. Content-Length: 0. Can you please let me know, what's going wrong and how can i proceed. Regards, Abhishek
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany
Hi Daniel,
As you instructed, i installed kamailio from the master branch (which has rtpengine module). Along with this, i installed the rtpengine package from sipwise, as instructed by them.
I also updated this param : modparam("nathelper", "sipping_from", " sip:pinger@abc.com") to my domain
Now the scenario is as follows:
1) I am able to call webrtc(firefox and chrome) from iphone, the signalling seems to be working fine, call can be paused, resumed etc.., but there is no audio/video transmission.
2) Still when i call from webrtc to iphone - the retpengine service of ubuntu terminates/crashes (like before) and needs to be restarted.
Does it have any thing to do with rtp port ranges? or is there some other misconfiguration?
Regards, Abhishek
On Tue, Sep 16, 2014 at 6:31 PM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
Hello,
maybe you should play with kamailio master branch (which is in testing phase before becoming 4.2) -- there you have the rtpengine -- and see if you get it working. Once that, you can look at using an older version, knowing you have it working and be able to compare. As I needed latest features, whenever I needed webrtc gatewaying, I used devel branch of rtpengine module.
Cheers, Daniel
On 16/09/14 14:24, Abhishek Saini wrote:
Hi Daniel,
I was able to solve a fraction of my problem, Actually, the github link had used rtpengine.so and i was using rptproxy-ng.so, there is a difference in the flag conventions between the two; i modified that to achieve a little progress.
Now, i am able to call on webrtc(firefox) from sip phone. However, after accepting call, there is no audio, and disconnecting the call from either end does not disconnect the call.
When i try to call from webrtc(firefox) to sip phone, there is no signalling at all, and the sip phone to webrtc calls can't connect after that. (I analyzed that mediaproxy-ng/rtpengine process terminates and has to be started again)
Following are the links to my latest kamailio.cfg file and port trace log of sip messages. http://jmp.sh/o0apKgP http://jmp.sh/HXnFRQj
I am clueless at the moment!
Regards, Abhishek
On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini < abhishek.saini@enukesoftware.com> wrote:
Hi Daniel,
Thanks for this.
I took the entire config files and configured it as per my ips and ports, after doing that, still no call establishment(webrtc to classic sip phones and vice-versa). Following is what i get in kamailio.log:
rtpp_test(): rtp proxy udp:127.0.0.1:7722 found, support for it enabled ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' ERROR: <script>: ==> duri=[ sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp] INFO: <script>: Request coming from WS ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' INFO: <script>: Reply from softphone: 100
And this SIP message: SIP/2.0 603 Failed to get local SDP.
Regards, Abhishek
On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
the reply code indicates that the media type is not supported, thus there has been no gatewaying between webrtc and classic rtp. Just replacing rtpproxy with rtpengine is not enough, there are different parameters that have to be provided.
Searching on web, I see that Carlos has published a config for it, see:
Cheers, Daniel
On 15/09/14 12:58, Abhishek Saini wrote:
Hi,
I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng package on my ubuntu box. As suggested here: http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module, but still not able to connect the webrtc calls to classic sip phones (and vice-versa). Below is the sip message that is traced:
SIP/2.0 488 Not acceptable here. Via: SIP/2.0/TCP 54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$ Via: SIP/2.0/WS df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$ From: "admin" sip:admin@abc.com;tag=bzhwwG8nT2gFwwJgIyrz. To: sip:hari@abc.com;tag=OIllTQf. Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a. CSeq: 65463 INVITE. User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2). Supported: replaces, outbound. Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards, Abhishek
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
Hi Daniel,
Here is something i traced in the log:
ip-172-31-47-138 rtpengine[4879]: Unknown flag encountered: 'force' ip-172-31-47-138 kernel: [4155571.651074] traps: rtpengine[4884] general protection ip:41e313 sp:7f2bf1934418 error:0 in rtpengine[400000+30000]
What's the cause of this error? i am using code from the master branch. Perhaps this has something to do with the rptengine service crash/termination.
Regards
On Wed, Sep 17, 2014 at 1:28 PM, Abhishek Saini < abhishek.saini@enukesoftware.com> wrote:
Hi Daniel,
As you instructed, i installed kamailio from the master branch (which has rtpengine module). Along with this, i installed the rtpengine package from sipwise, as instructed by them.
I also updated this param : modparam("nathelper", "sipping_from", " sip:pinger@abc.com") to my domain
Now the scenario is as follows:
- I am able to call webrtc(firefox and chrome) from iphone, the
signalling seems to be working fine, call can be paused, resumed etc.., but there is no audio/video transmission.
- Still when i call from webrtc to iphone - the retpengine service of
ubuntu terminates/crashes (like before) and needs to be restarted.
Does it have any thing to do with rtp port ranges? or is there some other misconfiguration?
Regards, Abhishek
On Tue, Sep 16, 2014 at 6:31 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
maybe you should play with kamailio master branch (which is in testing phase before becoming 4.2) -- there you have the rtpengine -- and see if you get it working. Once that, you can look at using an older version, knowing you have it working and be able to compare. As I needed latest features, whenever I needed webrtc gatewaying, I used devel branch of rtpengine module.
Cheers, Daniel
On 16/09/14 14:24, Abhishek Saini wrote:
Hi Daniel,
I was able to solve a fraction of my problem, Actually, the github link had used rtpengine.so and i was using rptproxy-ng.so, there is a difference in the flag conventions between the two; i modified that to achieve a little progress.
Now, i am able to call on webrtc(firefox) from sip phone. However, after accepting call, there is no audio, and disconnecting the call from either end does not disconnect the call.
When i try to call from webrtc(firefox) to sip phone, there is no signalling at all, and the sip phone to webrtc calls can't connect after that. (I analyzed that mediaproxy-ng/rtpengine process terminates and has to be started again)
Following are the links to my latest kamailio.cfg file and port trace log of sip messages. http://jmp.sh/o0apKgP http://jmp.sh/HXnFRQj
I am clueless at the moment!
Regards, Abhishek
On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini < abhishek.saini@enukesoftware.com> wrote:
Hi Daniel,
Thanks for this.
I took the entire config files and configured it as per my ips and ports, after doing that, still no call establishment(webrtc to classic sip phones and vice-versa). Following is what i get in kamailio.log:
rtpp_test(): rtp proxy udp:127.0.0.1:7722 found, support for it enabled ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' ERROR: <script>: ==> duri=[ sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp] INFO: <script>: Request coming from WS ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' INFO: <script>: Reply from softphone: 100
And this SIP message: SIP/2.0 603 Failed to get local SDP.
Regards, Abhishek
On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
the reply code indicates that the media type is not supported, thus there has been no gatewaying between webrtc and classic rtp. Just replacing rtpproxy with rtpengine is not enough, there are different parameters that have to be provided.
Searching on web, I see that Carlos has published a config for it, see:
Cheers, Daniel
On 15/09/14 12:58, Abhishek Saini wrote:
Hi,
I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng package on my ubuntu box. As suggested here: http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module, but still not able to connect the webrtc calls to classic sip phones (and vice-versa). Below is the sip message that is traced:
SIP/2.0 488 Not acceptable here. Via: SIP/2.0/TCP 54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$ Via: SIP/2.0/WS df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$ From: "admin" sip:admin@abc.com;tag=bzhwwG8nT2gFwwJgIyrz. To: sip:hari@abc.com;tag=OIllTQf. Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a. CSeq: 65463 INVITE. User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2). Supported: replaces, outbound. Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards, Abhishek
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
Hi,
I have reported a bug on rptengine github, for the crash issue: https://github.com/sipwise/rtpengine/issues/27
You mentioned that you have been using rtpengine kamailio module and the rtpengine debian package with success. Was it on ubuntu box or some other linux system? Sorry for asking this, but i am not able move ahead because of this. Any other module suggestions from your end?
Thanks
On Wed, Sep 17, 2014 at 6:45 PM, Abhishek Saini < abhishek.saini@enukesoftware.com> wrote:
Hi Daniel,
Here is something i traced in the log:
ip-172-31-47-138 rtpengine[4879]: Unknown flag encountered: 'force' ip-172-31-47-138 kernel: [4155571.651074] traps: rtpengine[4884] general protection ip:41e313 sp:7f2bf1934418 error:0 in rtpengine[400000+30000]
What's the cause of this error? i am using code from the master branch. Perhaps this has something to do with the rptengine service crash/termination.
Regards
On Wed, Sep 17, 2014 at 1:28 PM, Abhishek Saini < abhishek.saini@enukesoftware.com> wrote:
Hi Daniel,
As you instructed, i installed kamailio from the master branch (which has rtpengine module). Along with this, i installed the rtpengine package from sipwise, as instructed by them.
I also updated this param : modparam("nathelper", "sipping_from", " sip:pinger@abc.com") to my domain
Now the scenario is as follows:
- I am able to call webrtc(firefox and chrome) from iphone, the
signalling seems to be working fine, call can be paused, resumed etc.., but there is no audio/video transmission.
- Still when i call from webrtc to iphone - the retpengine service of
ubuntu terminates/crashes (like before) and needs to be restarted.
Does it have any thing to do with rtp port ranges? or is there some other misconfiguration?
Regards, Abhishek
On Tue, Sep 16, 2014 at 6:31 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
maybe you should play with kamailio master branch (which is in testing phase before becoming 4.2) -- there you have the rtpengine -- and see if you get it working. Once that, you can look at using an older version, knowing you have it working and be able to compare. As I needed latest features, whenever I needed webrtc gatewaying, I used devel branch of rtpengine module.
Cheers, Daniel
On 16/09/14 14:24, Abhishek Saini wrote:
Hi Daniel,
I was able to solve a fraction of my problem, Actually, the github link had used rtpengine.so and i was using rptproxy-ng.so, there is a difference in the flag conventions between the two; i modified that to achieve a little progress.
Now, i am able to call on webrtc(firefox) from sip phone. However, after accepting call, there is no audio, and disconnecting the call from either end does not disconnect the call.
When i try to call from webrtc(firefox) to sip phone, there is no signalling at all, and the sip phone to webrtc calls can't connect after that. (I analyzed that mediaproxy-ng/rtpengine process terminates and has to be started again)
Following are the links to my latest kamailio.cfg file and port trace log of sip messages. http://jmp.sh/o0apKgP http://jmp.sh/HXnFRQj
I am clueless at the moment!
Regards, Abhishek
On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini < abhishek.saini@enukesoftware.com> wrote:
Hi Daniel,
Thanks for this.
I took the entire config files and configured it as per my ips and ports, after doing that, still no call establishment(webrtc to classic sip phones and vice-versa). Following is what i get in kamailio.log:
rtpp_test(): rtp proxy udp:127.0.0.1:7722 found, support for it enabled ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' ERROR: <script>: ==> duri=[ sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp] INFO: <script>: Request coming from WS ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown option ` ' INFO: <script>: Reply from softphone: 100
And this SIP message: SIP/2.0 603 Failed to get local SDP.
Regards, Abhishek
On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
the reply code indicates that the media type is not supported, thus there has been no gatewaying between webrtc and classic rtp. Just replacing rtpproxy with rtpengine is not enough, there are different parameters that have to be provided.
Searching on web, I see that Carlos has published a config for it, see:
Cheers, Daniel
On 15/09/14 12:58, Abhishek Saini wrote:
Hi,
I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng package on my ubuntu box. As suggested here: http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
I have kept rtpproxy-ng's configuration same as the rtpproxy module, but still not able to connect the webrtc calls to classic sip phones (and vice-versa). Below is the sip message that is traced:
SIP/2.0 488 Not acceptable here. Via: SIP/2.0/TCP 54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$ Via: SIP/2.0/WS df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$ From: "admin" sip:admin@abc.com;tag=bzhwwG8nT2gFwwJgIyrz. To: sip:hari@abc.com;tag=OIllTQf. Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a. CSeq: 65463 INVITE. User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2). Supported: replaces, outbound. Content-Length: 0.
Can you please let me know, what's going wrong and how can i proceed.
Regards, Abhishek
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany