19 Dec
2006
19 Dec
'06
6:40 p.m.
Hello
tnx for the attention
At 17.00 19/12/2006, Stefano Capitanio wrote:
Hi,
is it correct that in the second Via Header of the message from
Asterisk to OpenSER there is the address of the Client?
there should be the address of Asterisk no?
I guess it is correct for outbound calls (from client to openser -> asterisk)
I captured some packets on a working setup and I see the same via
headers: one with OpenSER address and one with Client address
maybe this make OpenSER think that the message is
coming from a NAT...
i'm not sure, maybe i'm wrong...
try to set:
modparam("nathelper", "rtpproxy_disable", 1)
I also think it's a NAT-related issue but that modparam doesn't solve
the problem :(
Tnx again for help
Regards
Edoardo
regards,
Stefano
Edoardo Serra ha scritto:
>At 13.51 19/12/2006, Klaus Darilion wrote:
>>You said that the 200 contains openser's IP in the SDP? Is it put
>>in there by openser or already by Asterisk?
>
>Tnx very much for help
>
>It's put in there by OpenSER.
>
>I'm attaching the 2 SIP/SDP packets (1 from asterisk to openser and
>1 from openser to client)
>
>AAA.AAA.AAA.AAA stands for IP of Asterisk
>OOO.OOO.OOO.OOO stands for IP of OpenSER
>CCC.CCC.CCC.CCC stands for IP of client
>3333333333 is the called number
>
>
>No. Time Source Destination
>Protocol Info
> 20 12.646925 AAA.AAA.AAA.AAA OOO.OOO.OOO.OOO SIP/SDP
>Status: 200 OK, with session description
>
>Session Initiation Protocol
> Status-Line: SIP/2.0 200 OK
> Message Header
> Via: SIP/2.0/UDP
> OOO.OOO.OOO.OOO;branch=z9hG4bK5bbd.eaf4f093.0;received=OOO.OOO.OOO.OOO
> Via: SIP/2.0/UDP
> CCC.CCC.CCC.CCC:8952;branch=z9hG4bK-d87543-e15656230434101e-1--d87543-;rport=8952
>
> Record-Route: <sip:OOO.OOO.OOO.OOO;lr=on;ftag=9043ec70>
> From: "test"<sip:test@OOO.OOO.OOO.OOO>;tag=9043ec70
> To:
"3333333333"<sip:3333333333@OOO.OOO.OOO.OOO>;tag=as30a7528b
> Call-ID: 98684a222a2eeb7aYmVlZTUzZDRhNjMzN2Y0MTZhYmNmOTc5MzQ4OGI3ZGU.
> CSeq: 3 INVITE
> User-Agent: Asterisk
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:3333333333@AAA.AAA.AAA.AAA:5060>
> Content-Type: application/sdp
> Content-Length: 291
> Message body
> Session Description Protocol
> Session Description Protocol Version (v): 0
> Owner/Creator, Session Id (o): root 20137 20138 IN IP4
> AAA.AAA.AAA.AAA
> Session Name (s): session
> Connection Information (c): IN IP4 AAA.AAA.AAA.AAA
> Time Description, active time (t): 0 0
> Media Description, name and address (m): audio 58508
> RTP/AVP 98 3 8 0 101
> Media Attribute (a): rtpmap:98 iLBC/8000
> Media Attribute (a): rtpmap:3 GSM/8000
> Media Attribute (a): rtpmap:8 PCMA/8000
> Media Attribute (a): rtpmap:0 PCMU/8000
> Media Attribute (a): rtpmap:101 telephone-event/8000
> Media Attribute (a): fmtp:101 0-16
> Media Attribute (a): silenceSupp:off - - - -
>
>No. Time Source Destination
>Protocol Info
> 21 12.647437 OOO.OOO.OOO.OOO CCC.CCC.CCC.CCC SIP/SDP
>Status: 200 OK, with session description
>
>Session Initiation Protocol
> Status-Line: SIP/2.0 200 OK
> Message Header
> Via: SIP/2.0/UDP
> OOO.OOO.OOO.OOO:5060;branch=z9hG4bK-d87543-e15656230434101e-1--d87543-;rport=8952
>
> Record-Route: <sip:OOO.OOO.OOO.OOO;lr=on;ftag=9043ec70>
> From: "test"<sip:test@OOO.OOO.OOO.OOO>;tag=9043ec70
> To:
"3333333333"<sip:3333333333@OOO.OOO.OOO.OOO>;tag=as30a7528b
> Call-ID: 98684a222a2eeb7aYmVlZTUzZDRhNjMzN2Y0MTZhYmNmOTc5MzQ4OGI3ZGU.
> CSeq: 3 INVITE
> User-Agent: Asterisk
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:3333333333@AAA.AAA.AAA.AAA:5060>
> Content-Type: application/sdp
> Content-Length: 291
> Message body
> Session Description Protocol
> Session Description Protocol Version (v): 0
> Owner/Creator, Session Id (o): root 20137 20138 IN IP4
> OOO.OOO.OOO.OOO
> Session Name (s): session
> Connection Information (c): IN IP4 OOO.OOO.OOO.OOO
> Time Description, active time (t): 0 0
> Media Description, name and address (m): audio 58508
> RTP/AVP 98 3 8 0 101
> Media Attribute (a): rtpmap:98 iLBC/8000
> Media Attribute (a): rtpmap:3 GSM/8000
> Media Attribute (a): rtpmap:8 PCMA/8000
> Media Attribute (a): rtpmap:0 PCMU/8000
> Media Attribute (a): rtpmap:101 telephone-event/8000
> Media Attribute (a): fmtp:101 0-16
> Media Attribute (a): silenceSupp:off - - - -
>
>Tnx very much for help again
>
>Regards
>
>Edoardo
>
>
>>regards
>>klaus
>>
>>
>>regards
>>klaus
>>
>>Edoardo Serra wrote:
>>>Hi guys,
>>> I'm having a problem with an OpenSER acting as registrar
>>> server and load balancer for many Asterisk servers.
>>>In a few words: "users are registering on openser and, when they
>>>want to make a call, OpenSER proxies the request to an Asterisk
>>>server with the dispatcher module"
>>>Here is the intended data flow (SIP goes through OpenSER and
>>>media goes directly to Asterisk)
>>>User <-- SIP --> OpenSER <-- SIP --> Asterisk
>>>User <-- RTP --> Asterisk
>>>Both, OpenSER and Asterisks have public IPs
>>>I already have a working setup of that and everything seems
>>>working correctly.
>>>I'm trying to replicate that setup on another site, same
>>>configurations of the boxes, same versions of OpenSER and
>>>Asterisk, etc... but I'm having monodirectional Audio.
>>>Having a look with tethereal I see that OpenSER, when the
>>>communication is answered, sends a SIP packet (200 OK) to the
>>>user indicating itself as media endpoint instead of the Asterisks.
>>> From that moment I see RTP packets flowing from the client to OpenSER
>>>This seems really strange to me because I just copied the same
>>>configurations file from a working setup to the new installation.
>>>Tnx in advance for help.
>>>Regards
>>>P.S.: Here is my openser.cfg
>>>## $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
>>>## simple quick-start config script
>>>#
>>># ----------- global configuration parameters ------------------------
>>>#debug=3 # debug level (cmd line: -dddddddddd)
>>>fork=yes
>>>#log_stderror=no # (cmd line: -E)
>>>check_via=no # (cmd. line: -v)
>>>dns=no # (cmd. line: -r)
>>>rev_dns=no # (cmd. line: -R)
>>>#children=4
>>>#port=5060
>>>fifo="/tmp/ser_fifo"
>>>#uid=nobody
>>>#gid=nobody
>>># ------------------ module loading ----------------------------------
>>>loadmodule "/usr/lib/openser/modules/sl.so"
>>>loadmodule "/usr/lib/openser/modules/tm.so"
>>>loadmodule "/usr/lib/openser/modules/rr.so"
>>>loadmodule "/usr/lib/openser/modules/maxfwd.so"
>>>loadmodule "/usr/lib/openser/modules/usrloc.so"
>>>loadmodule "/usr/lib/openser/modules/registrar.so"
>>>loadmodule "/usr/lib/openser/modules/nathelper.so"
>>>loadmodule "/usr/lib/openser/modules/textops.so"
>>>loadmodule "/usr/lib/openser/modules/exec.so"
>>>loadmodule "/usr/lib/openser/modules/uri.so"
>>>loadmodule "/usr/lib/openser/modules/uri_db.so"
>>>loadmodule "/usr/lib/openser/modules/dispatcher.so"
>>># Uncomment this if you want digest authentication
>>># mysql.so must be loaded !
>>>loadmodule "/usr/lib/openser/modules/mysql.so"
>>>loadmodule "/usr/lib/openser/modules/auth.so"
>>>loadmodule "/usr/lib/openser/modules/auth_db.so"
>>>modparam("usrloc", "db_mode", 2)
>>>modparam("usrloc", "db_url",
"mysql://xxx:xxx@xxx.xxx.xxx.xxx/openser")
>>>modparam("usrloc", "timer_interval", 120)
>>>modparam("auth_db", "calculate_ha1", 0)
>>>modparam("auth_db", "db_url",
"mysql://xxx:xxx@xxx.xxx.xxx.xxx/voip")
>>>modparam("uri_db", "db_url",
"mysql://xxx:xxx@xxx.xxx.xxx.xxx/openser")
>>>modparam("rr", "enable_full_lr", 1)
>>>modparam("registrar", "nat_flag", 6)
>>>modparam("registrar", "max_expires", 3600)
>>>modparam("registrar", "min_expires", 60)
>>>modparam("registrar", "append_branches", 0)
>>>modparam("registrar", "desc_time_order", 1)
>>>modparam("nathelper", "natping_interval", 20) # Ping
interval 20 s
>>>modparam("nathelper", "ping_nated_only", 1) # Ping only
clients behind NAT
>>>modparam("dispatcher", "force_dst", 1)
>>># ------------------------- request routing logic -------------------
>>># main routing logic
>>>route{
>>># initial sanity checks -- messages with
>>># max_forwards==0, or excessively long requests
>>>if (!mf_process_maxfwd_header("10")) {
>>>sl_send_reply("483","Too Many Hops");
>>>exit;
>>>};
>>>if ( msg:len > max_len ) {
>>>sl_send_reply("513", "Message too big");
>>>exit;
>>>};
>>>if ( (method=="OPTIONS") || (method=="SUBSCRIBE") ||
(method=="NOTIFY") ) {
>>>sl_send_reply("405", "Method Not Allowed");
>>>exit;
>>>}
>>>if (!method=="REGISTER") {
>>>record_route();
>>>};
>>>if ((src_ip==xxx.xxx.xxx.xxx) || (src_ip==xxx.xxx.xxx.xxx)) { #
>>>IP of Asterisks
>>>if (!lookup("location")) {
>>>sl_send_reply("404", "Not Found");
>>>exit;
>>>};
>>># forward to current uri now; use stateful forwarding; that
>>># works reliably even if we forward from TCP to UDP
>>>if (!t_relay()) {
>>>sl_reply_error();
>>>};
>>>exit;
>>>};
>>>if (nat_uac_test("3")) {
>>>if ((method=="REGISTER") || (method=="INVITE") ||
(method=="OPTIONS")) {
>>>fix_nated_contact();
>>>force_rport();
>>>setflag(6); # Mark as NATed
>>>}
>>>}
>>># if the request is for other domain use UsrLoc
>>># (in case, it does not work, use the following command
>>># with proper names and addresses in it)
>>>if (method=="REGISTER") {
>>>if (!proxy_authorize("domain", "openser_view")) {
>>>proxy_challenge("domain", "0");
>>>exit;
>>>}
>>>if (!check_to()) {
>>>sl_send_reply("403", "Digest username and URI username do NOT
>>>match! Stay away!");
>>>exit;
>>>}
>>>save("location");
>>>exit;
>>>};
>>>
>>>if (method=="INVITE") {
>>>if (!proxy_authorize("domain", "openser_view")) {
>>>proxy_challenge("domain", "0");
>>>exit;
>>>}
>>>if (!check_from()) {
>>>sl_send_reply("403", "Digest username and URI username do NOT
>>>match! Stay away!");
>>>exit;
>>>}
>>>}
>>># loose-route processing
>>>if (loose_route()) {
>>># mark routing logic in request
>>>append_hf("P-hint: rr-enforced\r\n");
>>>route(1);
>>>exit;
>>>};
>>>if (!uri==myself) {
>>># mark routing logic in request
>>>append_hf("P-hint: outbound\r\n");
>>>route(1);
>>>exit;
>>>};
>>>append_hf("P-hint: usrloc applied\r\n");
>>>route(1);
>>>}
>>>route[1]
>>>{
>>># ! Nathelper
>>>if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
&&
>>>!search("^Route:")){
>>>sl_send_reply("479", "We don't forward to private IP
addresses");
>>>exit;
>>>};
>>># NAT processing of replies; apply to all transactions (for example,
>>># re-INVITEs from public to private UA are hard to identify as
>>># NATed at the moment of request processing); look at replies
>>>t_on_reply("1");
>>># send it out now; use stateful forwarding as it works reliably
>>># even for UDP2TCP
>>>if ((src_ip!=xxx.xxx.xxx.xxx) && (src_ip!=xxx.xxx.xxx.xxx)) { #
>>>IP of Asterisks
>>>ds_select_dst("2", "0");
>>>}
>>>if (!t_relay()) {
>>>sl_reply_error();
>>>};
>>>}
>>># ! Nathelper
>>>onreply_route[1] {
>>># NATed transaction ?
>>>if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
>>>fix_nated_contact();
>>># otherwise, is it a transaction behind a NAT and we did not
>>># know at time of request processing ? (RFC1918 contacts)
>>>} else if (nat_uac_test("1")) {
>>>fix_nated_contact();
>>>};
>>>}
>>>
>>>_______________________________________________
>>>Users mailing list
>>>Users(a)openser.org
>>>http://openser.org/cgi-bin/mailman/listinfo/users
>>
>>
>>--
>>Klaus Darilion
>>nic.at
>
>
>_______________________________________________
>Users mailing list
>Users(a)openser.org
>http://openser.org/cgi-bin/mailman/listinfo/users