This is how I configured rtpproxy. Not sure why rtpproxy is not engaged both ways. Thanks for your help.

root@Kamailio:/etc/default# more rtpproxy # Defaults for rtpproxy

# The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line:

LISTEN_ADDR=public ip address CONTROL_SOCK="udp:localhost:22222"

# Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" root@Kamailio:/etc/default#

##kamailio.cfg##

#!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif

# ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222")

# ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@mydomain.com")

# params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif

# Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }

route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif

# RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; }

On Fri, Jun 3, 2011 at 4:33 AM, Alex Balashov abalashov@evaristesys.com wrote:

This sounds like an asymmetrical engagement of rtpproxy (e.g. on the reply but not the initial request, or vice versa).

On 06/02/2011 09:45 PM, Mokhtar Bengana wrote:

...

I have Kamailio and rtpproxy running behind NAT on the same server. Inbound calls to Kamailio from the outside work fine and I have two way audio but when I try an outbound echo test call the call connects but I have no audio. rtpproxy seems to be running and all rtp ports are poiting to Kamailio internal ip. The syslog is showing this error. Not sure what's missing in the config. Appreciate any inputs. Thanks.

root@Kamailio:~# tail /var/log/syslog Jun  3 01:24:02 Kamailio kamailio[810]: INFO: rtpproxy [rtpproxy.c:1403]: rtp proxyudp:127.0.0.1:22222  found, support for it enabled Jun  3 01:25:05 Kamailio kamailio[791]: INFO:<core>  [forward.c:786]: broken reply to forward - no 2nd via Jun  3 01:25:29 Kamailio kamailio[791]: ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy Jun  3 01:25:30 Kamailio kamailio[793]: ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy

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-- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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Andrew,

I am trying to use rtpproxy_offer/answer on the ONREPLY ROUTE but I am having some syntax issues. Here is the example I am using. Any help with the syntax appreciated.

route { ... if (is_method("INVITE")) { if (has_sdp()) { if (rtpproxy_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } if (is_method("ACK") && has_sdp()) rtpproxy_answer(); ... }

onreply_route[1] { ... if (has_sdp()) rtpproxy_answer(); ... }

onreply_route[2] { ... if (has_sdp()) rtpproxy_offer(); ... }

On Mon, Jun 6, 2011 at 5:37 AM, Andrew Pogrebennyk andrew.pogrebennyk@portaone.com wrote:

Mokhtar, could you please make sure that you are calling route(RTPPROXY) in reply route as well, as Alex suggested?

On 03.06.2011 16:33, Mokhtar Bengana wrote:

...

This is how I configured rtpproxy. Not sure why rtpproxy is not engaged both ways. Thanks for your help.

-- Sincerely, Andrew Pogrebennyk

---

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

I am still unable to fix the one way audio for my outgoing calls. I tried different flags for nat_uac_test and force_rtp_proxy and nothing made a difference. Wireshack is not showing an rtp stream being initiated from the callee back to my natted caller ip. I have included my routing logic and appreciate any help figuring out why rtpproxy is not engaged in both legs of the rtp stream.

# - 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

#!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222")

# ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@mydomain.com")

# params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endi

####### Routing Logic ########

# Main SIP request routing logic # - processing of any incoming SIP request starts with this route route {

# per request initial checks route(REQINIT);

# NAT detection route(NAT);

# handle requests within SIP dialogs route(WITHINDLG);

### only initial requests (no To tag)

# CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; }

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); }

route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif

/* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); }

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()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } 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 t_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_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif

if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } }

# when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } }

# Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return;

#!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 if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; }

if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else {

#!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif

# authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } }

consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; }

# Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("31")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }

# RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy("s"); } if (!has_totag()) add_rr_param(";nat=yes"); #!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; }

$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);

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

# Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); }

# Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif }

# Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif

if (t_is_canceled()) { exit; }

# uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##}

# uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

On Mon, Jun 6, 2011 at 10:52 PM, Alex Balashov abalashov@evaristesys.com wrote:

What are the syntax issues?

P.S. has_sdp() doesn't actually exist.

-- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

On Jun 6, 2011, at 11:48 PM, Mokhtar Bengana mokhtar.bengana@gmail.com wrote:

...

Andrew,

I am trying to use rtpproxy_offer/answer on the ONREPLY ROUTE but I am having some syntax issues. Here is the example I am using. Any help with the syntax appreciated.

route { ...    if (is_method("INVITE")) {        if (has_sdp()) {            if (rtpproxy_offer())                t_on_reply("1");        } else {            t_on_reply("2");        }    }    if (is_method("ACK") && has_sdp())        rtpproxy_answer(); ... }

onreply_route[1] { ...    if (has_sdp())        rtpproxy_answer(); ... }

onreply_route[2] { ...    if (has_sdp())        rtpproxy_offer(); ... }

On Mon, Jun 6, 2011 at 5:37 AM, Andrew Pogrebennyk andrew.pogrebennyk@portaone.com wrote:

...

Mokhtar, could you please make sure that you are calling route(RTPPROXY) in reply route as well, as Alex suggested?

On 03.06.2011 16:33, Mokhtar Bengana wrote:

...

This is how I configured rtpproxy. Not sure why rtpproxy is not engaged both ways. Thanks for your help.

-- Sincerely, Andrew Pogrebennyk

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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

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