[OpenSIPS-Users] No ACK response for 200 ok
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Wed Dec 22 11:44:24 CET 2010
Hi Nawfel,
Try to do fix_nated_sdp("1") (see
http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id293148)
to force Direction Active on Cisco GW (when sending a reply / request to
Cisco)
Regards,
Bogdan
Nawfel Oujdi wrote:
> Sorry Bogdan but now my setup become a bit differente, i have the same
> servers , Opensips+Asterisk in EC2 amazon (same LAN) and Cisco
> gateway outside conected through public_ip to Opensips.
> The SIP signalling works well but i have just oneway audio cause
> asterisk send private ip on the reply to opensips invite (in same
> LAN) and opensips forward that private ip to Cisco. So asterisk
> know the public ip of cisco to establish rtp traffic but cisco don´t.
> ¿how can i solve this problem ? ¿there is anyway to change the rtp ip
> in the invite's reply ?
> Best Regards!!
>
>
> opensips.cfg:
> route{
>
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","looping");
> exit;
> }
> if ($rU==NULL) {
> sl_send_reply("484","Address Incomplete");
> exit;
> }
> if (!has_totag()) {
> record_route_preset(" Opensips public ip ");
> xlog("route recorded \n");
> } else {
> loose_route();
> t_relay();
> exit;
> }
> if ( is_method("CANCEL") ) {
> if ( t_check_trans() )
> t_relay();
> exit;
> }
> if (!is_method("INVITE")) {
> send_reply("405","Method Not Allowed");
> exit;
> }
> if (method=="INVITE") {
> load_balance("1","calls");
> }
>
> if ($retcode<0) {
> sl_send_reply("500","Service full");
> exit;
> }
>
> xlog("Selected destination is: $du\n");
>
> if (!t_relay()) {
> sl_reply_error();
> }
> }
>
> ######################################################################################################
>
> U 2010/12/03 13:00:27.034603 80.65.13.238:65071
> <http://80.65.13.238:65071> -> 10.229.123.198:5060
> <http://10.229.123.198:5060>
> INVITE sip:911126667 at x.911126667.opensips.lab.egtelecom.es:5060
> <http://sip:911126667@x.911126667.opensips.lab.egtelecom.es:5060> SIP/2.0.
> Date: Fri, 03 Dec 2010 12:04:32 GMT.
> Call-Info: <sip:80.65.13.238:5060
> <http://80.65.13.238:5060>>;method="NOTIFY;Event=telephone-event;Duration=2000".
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY, INFO, REGISTER.
> From: <sip:911873699 at 80.65.13.238
> <mailto:sip%3A911873699 at 80.65.13.238>>;tag=274FBBA0-208D.
> Allow-Events: telephone-event.
> Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
> Min-SE: 1800.
> Remote-Party-ID: <sip:911873699 at 80.65.13.238
> <mailto:sip%3A911873699 at 80.65.13.238>>;party=calling;screen=yes;privacy=off.
> Cisco-Guid: 1378169425-4262203871-3197108258-2438471722.
> Timestamp: 1291377872.
> Content-Length: 269.
> User-Agent: Cisco-SIPGateway/IOS-12.x.
> To: <sip:911126667 at x.911126667.opensips.lab.egtelecom.es
> <mailto:sip%3A911126667 at x.911126667.opensips.lab.egtelecom.es>>.
> Contact: <sip:911873699 at 80.65.13.238:5060
> <http://sip:911873699@80.65.13.238:5060>>.
> Expires: 180.
> Content-Disposition: session;handling=required.
> Content-Type: application/sdp.
> Call-ID: 5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238
> <mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238>.
> Via: SIP/2.0/UDP
> 80.65.13.238:5060;x-route-tag="cid:Orange at 80.65.13.238
> <mailto:cid%3AOrange at 80.65.13.238>";branch=z9hG4bK1EDAC71C9A.
> CSeq: 101 INVITE.
> Max-Forwards: 70.
> .
> v=0.
> o=CiscoSystemsSIP-GW-UserAgent 849 9795 IN IP4 80.65.13.238.
> s=SIP Call.
> c=IN IP4 80.65.13.238.
> t=0 0.
> m=audio 23660 RTP/AVP 18 101.
> c=IN IP4 80.65.13.238.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2010/12/03 13:00:27.035190 10.229.123.198:5060
> <http://10.229.123.198:5060> -> 80.65.13.238:5060
> <http://80.65.13.238:5060>
> SIP/2.0 100 Giving a try.
> From: <sip:911873699 at 80.65.13.238
> <mailto:sip%3A911873699 at 80.65.13.238>>;tag=274FBBA0-208D.
> To: <sip:911126667 at x.911126667.opensips.lab.egtelecom.es
> <mailto:sip%3A911126667 at x.911126667.opensips.lab.egtelecom.es>>.
> Call-ID: 5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238
> <mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238>.
> Via: SIP/2.0/UDP
> 80.65.13.238:5060;x-route-tag="cid:Orange at 80.65.13.238
> <mailto:cid%3AOrange at 80.65.13.238>";branch=z9hG4bK1EDAC71C9A.
> CSeq: 101 INVITE.
> Server: OpenSIPS (1.6.3-notls (i386/linux)).
> Content-Length: 0.
> .
>
>
> U 2010/12/03 13:00:27.035263 10.229.123.198:5060
> <http://10.229.123.198:5060> -> 10.228.26.150:5060
> <http://10.228.26.150:5060>
> INVITE sip:911126667 at x.911126667.opensips.lab.egtelecom.es:5060
> <http://sip:911126667@x.911126667.opensips.lab.egtelecom.es:5060> SIP/2.0.
> Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.
> Date: Fri, 03 Dec 2010 12:04:32 GMT.
> Call-Info: <sip:80.65.13.238:5060
> <http://80.65.13.238:5060>>;method="NOTIFY;Event=telephone-event;Duration=2000".
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY, INFO, REGISTER.
> From: <sip:911873699 at 80.65.13.238
> <mailto:sip%3A911873699 at 80.65.13.238>>;tag=274FBBA0-208D.
> Allow-Events: telephone-event.
> Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
> Min-SE: 1800.
> Remote-Party-ID: <sip:911873699 at 80.65.13.238
> <mailto:sip%3A911873699 at 80.65.13.238>>;party=calling;screen=yes;privacy=off.
> Cisco-Guid: 1378169425-4262203871-3197108258-2438471722.
> Timestamp: 1291377872.
> Content-Length: 269.
> User-Agent: Cisco-SIPGateway/IOS-12.x.
> To: <sip:911126667 at x.911126667.opensips.lab.egtelecom.es
> <mailto:sip%3A911126667 at x.911126667.opensips.lab.egtelecom.es>>.
> Contact: <sip:911873699 at 80.65.13.238:5060
> <http://sip:911873699@80.65.13.238:5060>>.
> Expires: 180.
> Content-Disposition: session;handling=required.
> Content-Type: application/sdp.
> Call-ID: 5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238
> <mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238>.
> Via: SIP/2.0/UDP 46.51.135.212;branch=z9hG4bK1a6f.13422624.0.
> Via: SIP/2.0/UDP
> 80.65.13.238:5060;x-route-tag="cid:Orange at 80.65.13.238
> <mailto:cid%3AOrange at 80.65.13.238>";branch=z9hG4bK1EDAC71C9A.
> CSeq: 101 INVITE.
> Max-Forwards: 69.
> .
> v=0.
> o=CiscoSystemsSIP-GW-UserAgent 849 9795 IN IP4 80.65.13.238.
> s=SIP Call.
> c=IN IP4 80.65.13.238.
> t=0 0.
> m=audio 23660 RTP/AVP 18 101.
> c=IN IP4 80.65.13.238.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2010/12/03 13:00:27.036250 10.228.26.150:5060
> <http://10.228.26.150:5060> -> 10.229.123.198:5060
> <http://10.229.123.198:5060>
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP
> 46.51.135.212;branch=z9hG4bK1a6f.13422624.0;received=10.229.123.198.
> Via: SIP/2.0/UDP
> 80.65.13.238:5060;x-route-tag="cid:Orange at 80.65.13.238
> <mailto:cid%3AOrange at 80.65.13.238>";branch=z9hG4bK1EDAC71C9A.
> Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.
> From: <sip:911873699 at 80.65.13.238
> <mailto:sip%3A911873699 at 80.65.13.238>>;tag=274FBBA0-208D.
> To: <sip:911126667 at x.911126667.opensips.lab.egtelecom.es
> <mailto:sip%3A911126667 at x.911126667.opensips.lab.egtelecom.es>>.
> Call-ID: 5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238
> <mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238>.
> CSeq: 101 INVITE.
> Server: Asterisk PBX 1.6.2.13.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
> Supported: replaces, timer.
> Require: timer.
> Session-Expires: 1800;refresher=uas.
> Contact: <sip:911126667 at 10.228.26.150
> <mailto:sip%3A911126667 at 10.228.26.150>>.
> Content-Length: 0.
> .
>
>
> U 2010/12/03 13:00:27.235884 10.228.26.150:5060
> <http://10.228.26.150:5060> -> 10.229.123.198:5060
> <http://10.229.123.198:5060>
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 46.51.135.212;branch=z9hG4bK1a6f.13422624.0;received=10.229.123.198.
> Via: SIP/2.0/UDP
> 80.65.13.238:5060;x-route-tag="cid:Orange at 80.65.13.238
> <mailto:cid%3AOrange at 80.65.13.238>";branch=z9hG4bK1EDAC71C9A.
> Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.
> From: <sip:911873699 at 80.65.13.238
> <mailto:sip%3A911873699 at 80.65.13.238>>;tag=274FBBA0-208D.
> To: <sip:911126667 at x.911126667.opensips.lab.egtelecom.es
> <mailto:sip%3A911126667 at x.911126667.opensips.lab.egtelecom.es>>;tag=as33981ab2.
> Call-ID: 5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238
> <mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238>.
> CSeq: 101 INVITE.
> Server: Asterisk PBX 1.6.2.13.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
> Supported: replaces, timer.
> Require: timer.
> Session-Expires: 1800;refresher=uas.
> Contact: <sip:911126667 at 10.228.26.150
> <mailto:sip%3A911126667 at 10.228.26.150>>.
> Content-Type: application/sdp.
> Content-Length: 262.
> .
> v=0.
> o=root 1270939673 1270939673 IN IP4 10.228.26.150.
> s=Asterisk PBX 1.6.2.13.
> c=IN IP4 10.228.26.150.
> t=0 0.
> m=audio 10532 RTP/AVP 18 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
>
> U 2010/12/03 13:00:27.236908 10.229.123.198:5060
> <http://10.229.123.198:5060> -> 80.65.13.238:5060
> <http://80.65.13.238:5060>
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 80.65.13.238:5060;x-route-tag="cid:Orange at 80.65.13.238
> <mailto:cid%3AOrange at 80.65.13.238>";branch=z9hG4bK1EDAC71C9A.
> Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.
> From: <sip:911873699 at 80.65.13.238
> <mailto:sip%3A911873699 at 80.65.13.238>>;tag=274FBBA0-208D.
> To: <sip:911126667 at x.911126667.opensips.lab.egtelecom.es
> <mailto:sip%3A911126667 at x.911126667.opensips.lab.egtelecom.es>>;tag=as33981ab2.
> Call-ID: 5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238
> <mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238>.
> CSeq: 101 INVITE.
> Server: Asterisk PBX 1.6.2.13.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
> Supported: replaces, timer.
> Require: timer.
> Session-Expires: 1800;refresher=uas.
> Contact: <sip:911126667 at 10.228.26.150
> <mailto:sip%3A911126667 at 10.228.26.150>>.
> Content-Type: application/sdp.
> Content-Length: 262.
> .
> v=0.
> o=root 1270939673 1270939673 IN IP4 10.228.26.150.
> s=Asterisk PBX 1.6.2.13.
> c=IN IP4 10.228.26.150.
> t=0 0.
> m=audio 10532 RTP/AVP 18 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
>
> U 2010/12/03 13:00:27.294728 80.65.13.238:65071
> <http://80.65.13.238:65071> -> 10.229.123.198:5060
> <http://10.229.123.198:5060>
> ACK sip:911126667 at 10.228.26.150:5060
> <http://sip:911126667@10.228.26.150:5060> SIP/2.0.
> Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.
> Date: Fri, 03 Dec 2010 12:04:32 GMT.
> From: <sip:911873699 at 80.65.13.238
> <mailto:sip%3A911873699 at 80.65.13.238>>;tag=274FBBA0-208D.
> Allow-Events: telephone-event.
> Content-Length: 0.
> To: <sip:911126667 at x.911126667.opensips.lab.egtelecom.es
> <mailto:sip%3A911126667 at x.911126667.opensips.lab.egtelecom.es>>;tag=as33981ab2.
> Call-ID: 5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238
> <mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238>.
> Via: SIP/2.0/UDP
> 80.65.13.238:5060;x-route-tag="cid:Orange at 80.65.13.238
> <mailto:cid%3AOrange at 80.65.13.238>";branch=z9hG4bK1EDAC8B3D.
> CSeq: 101 ACK.
> Max-Forwards: 70.
> .
>
>
> U 2010/12/03 13:00:27.295705 10.229.123.198:5060
> <http://10.229.123.198:5060> -> 10.228.26.150:5060
> <http://10.228.26.150:5060>
> ACK sip:911126667 at 10.228.26.150:5060
> <http://sip:911126667@10.228.26.150:5060> SIP/2.0.
> Date: Fri, 03 Dec 2010 12:04:32 GMT.
> From: <sip:911873699 at 80.65.13.238
> <mailto:sip%3A911873699 at 80.65.13.238>>;tag=274FBBA0-208D.
> Allow-Events: telephone-event.
> Content-Length: 0.
> To: <sip:911126667 at x.911126667.opensips.lab.egtelecom.es
> <mailto:sip%3A911126667 at x.911126667.opensips.lab.egtelecom.es>>;tag=as33981ab2.
> Call-ID: 5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238
> <mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE at 80.65.13.238>.
> Via: SIP/2.0/UDP 46.51.135.212;branch=z9hG4bK1a6f.13422624.2.
> Via: SIP/2.0/UDP
> 80.65.13.238:5060;x-route-tag="cid:Orange at 80.65.13.238
> <mailto:cid%3AOrange at 80.65.13.238>";branch=z9hG4bK1EDAC8B3D.
> CSeq: 101 ACK.
> Max-Forwards: 69.
>
> 2010/12/2 Bogdan-Andrei Iancu <bogdan at voice-system.ro
> <mailto:bogdan at voice-system.ro>>
>
> Hi Nawfel,
>
> The problem is in one of the end points as for a 200 OK calls, the
> 200 reply and the ACK is end-2-end.
>
> If you have a trace, maybe I can help you to see if there is a
> signalling problem.
>
> Regards,
> Bogdan
>
>
> Nawfel Oujdi wrote:
>
> Hello!!
> I m new in opensips and i m testing the load balancer cause i
> need it to balance calls between 4 asterisk.For the start i
> make the following scenario
> Cisco gateway inbound ------> opensips ------> asterisk
> ---------> Cisco gateway outbound
> when the call comes to the opensips, the load_balancer
> forward the call correctly to my asterisk but the call hangs
> up after 15 seg approximately.When i did a ngrep for the sip
> traffic in opensips, i realized that cisco gateway inbound
> never sent the ACK for 200 OK to opensips .
> In the Cisco's logs i saw that the reply of 200 ok is sent
> directly to public ip of asterisk but never to opensips server
> so asterisk still waiting for the ACK from opensips.
> In the same way opensips never receive the BYE packet and the
> load never decrease when the call is hanging up.
>
> Cisco gateway opensips asterisk
> ---invite--->
> <--trying---- ---invite--->
> <---trying---
> <----200OK---
> <---200 OK---
> <----200OK---
> <---200 OK---
> <----200OK---
> <---200 OK---
> <----200OK---
> <---200 OK---
> Please can somebady help me to understand what cause that?
>
> Best Regards!!
>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 15 - 19 November 2010, Edison, New Jersey, USA
> www.voice-system.ro <http://www.voice-system.ro>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> --
>
> Nawfel Oujdi
> *Ingeniero VoIP*
> noujdi at egtelecom.es <mailto:noujdi at egtelecom.es>
> EG telecom S.A | www.egtelecom.es <http://www.egtelecom.es/>
> Oficina: *902 050 080*
> Agustín de Foxá, 25 - 9B | 28036 Madrid
>
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--
Bogdan-Andrei Iancu
OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami, USA
www.voice-system.ro
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