You need to use "ie" on offer and "ei" on answer on one call direction.
For the opposite call direction, use "ei" on offer an "ie" on answer.

Regards,
Ovidiu Sas

On Thu, Apr 4, 2013 at 12:24 PM, OCEANET - Cédric BASSAGET
<[email protected]> wrote:
> Hello,
>
> For my first post on this mailing list, I'm trying to make opensips working
> in this scheme (that's a lab test, so don't take care of my "public" ip
> addresses) :
>
> UAC (asterisk) <---------------------> routeur(nat) <------------------>
> opensips (mhomed) <----------> sip privoder
> 10.10.1.1                               10.10.1.254 / 10.7.59.1
> 10.7.59.2 / 10.0.95.10                  10.0.95.1
>
> mhomed is set to 1
> I have a rtpproxy set in bridge mode on my opensips, like this :
> /usr/local/bin/rtpproxy -p /var/run/rtpproxy.pid -u rtpproxy -d DBUG
> LOG_LOCAL5 -m 10000 -M 20000 -l 10.0.95.10 10.7.59.2 -s udp:127.0.0.1 9999
>
> I'm trying to establish a call from my asterisk to a public phone number,
> but I have a problem with SDP reply (200 OK) , from my opensips to my UAC.
> SDP c= value is still 10.0.95.10, instead of 10.7.59.2. That's the only
> thing that is not correct in the SIP session. I can't find why c= value is
> not correct in the 200 OK.
>
> I use engage_rtp_proxy("fr"), but I tried with many combinations
> (rtpproxy_offer / answer, params=fier, params=feir).
>
> If somebody has already been in this situation, I would appreciate his help.
> Thanks for your replies, please tell me if you need more captures or
> anything else
>
> Here's a short description of SIP requests, seen from OpenSIPS :
>
>
> ##### INVITE FROM MY UAC TO OPENSIPS ###############
> U 2013/04/04 18:18:09.621724 10.7.59.1:1028 -> 10.7.59.2:5060
> INVITE sip:[email protected] SIP/2.0
> Via: SIP/2.0/UDP 10.7.59.1:1029;branch=z9hG4bK13253a27;rport
> Max-Forwards: 70
> From: "100" <sip:[email protected]:1029>;tag=as747d5e5b
> To: <sip:[email protected]>
> Contact: <sip:[email protected]:1029>
> Call-ID: [email protected]:5060
> CSeq: 102 INVITE
> User-Agent: Oceanet-test
> Date: Thu, 04 Apr 2013 16:12:34 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 282
>
> v=0
> o=root 1416933599 1416933599 IN IP4 10.10.1.1
> s=Asterisk PBX 1.8.20.1
> c=IN IP4 10.7.59.1
> t=0 0
> m=audio 17282 RTP/AVP 9 0 101
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> #####INVITE FROM OPENSIPS TO PROVIDER ###############
> U 2013/04/04 18:18:09.622378 10.0.95.10:5060 -> 10.0.95.1:5060
> INVITE sip:[email protected] SIP/2.0
> Via: SIP/2.0/UDP 10.0.95.10;branch=z9hG4bKbd97.ced0f0b2.0
> Max-Forwards: 69
> From: "100" <sip:[email protected]:1029>;tag=as747d5e5b
> To: <sip:[email protected]>
> Contact: <sip:10.0.95.10;did=b0b.7e82f333>
> Call-ID: [email protected]:5060
> CSeq: 102 INVITE
> User-Agent: Oceanet-test
> Date: Thu, 04 Apr 2013 16:12:34 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 283
>
> v=0
> o=root 1416933599 1416933599 IN IP4 10.10.1.1
> s=Asterisk PBX 1.8.20.1
> c=IN IP4 10.0.95.10
> t=0 0
> m=audio 19392 RTP/AVP 9 0 101
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ##### REPLY FROM PROVIDER TO OPENSIPS ###############
> U 2013/04/04 18:18:16.327567 10.0.95.1:5060 -> 10.0.95.10:5060
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.95.10;branch=z9hG4bKbd97.ced0f0b2.0
> Call-ID: [email protected]:5060
> From: "100" <sip:[email protected]:1029>;tag=as747d5e5b
> To: <sip:[email protected]>;tag=a94c095b773be1dd6e8d668a785a9c84530b7132
> Contact: <sip:10.0.95.1;did=b0b.c3c431b5>
> CSeq: 102 INVITE
> Server: Dialogic-SIP/10.5.3.360 ACKBAR 0
> Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO,
> REFER, UPDATE
> Supported: path, replaces, timer, tdialog
> Accept: application/sdp, application/dtmf-relay, text/plain
> Content-Type: application/sdp
> Content-Length: 238
>
> v=0
> o=Dialogic_SDP 2521466 0 IN IP4 91.213.145.116
> s=Dialogic-SIP
> c=IN IP4 10.0.95.3
> t=0 0
> m=audio 10018 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=silenceSupp:off - - - -
> a=ptime:20
>
> #### REPLY FROM OPENSIPS TO UAC WITH WRONG SDP ################
> U 2013/04/04 18:18:16.328296 10.7.59.2:5060 -> 10.7.59.1:1028
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.7.59.1:1029;branch=z9hG4bK13253a27;rport
> Call-ID: [email protected]:5060
> From: "100" <sip:[email protected]:1029>;tag=as747d5e5b
> To: <sip:[email protected]>;tag=a94c095b773be1dd6e8d668a785a9c84530b7132
> Contact: <sip:10.7.59.2;did=b0b.7e82f333>
> CSeq: 102 INVITE
> Server: Dialogic-SIP/10.5.3.360 ACKBAR 0
> Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO,
> REFER, UPDATE
> Supported: path, replaces, timer, tdialog
> Accept: application/sdp, application/dtmf-relay, text/plain
> Content-Type: application/sdp
> Content-Length: 239
>
> v=0
> o=Dialogic_SDP 2521466 0 IN IP4 91.213.145.116
> s=Dialogic-SIP
> c=IN IP4 10.0.95.10
> t=0 0
> m=audio 11526 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=silenceSupp:off - - - -
> a=ptime:20
>
>
> Cédric
>
> _______________________________________________
> Users mailing list
> [email protected]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
VoIP Embedded, Inc.
http://www.voipembedded.com

_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Reply via email to