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I have a SIP provider: telecomitalia that use a cirpack! The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I don't know if it follows RFC...)
When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session description is modified. Both TRYING/RINGING with session description Connection Address/port are not replaced!
OpenWengo understand the modification and change the IP where it sends RTP from the one in TRYING/RINGING SDP to the one in 200ok (proxy address because of force_rtp_proxy()) But other softphones like Twinkle don't understand it and still send RTP to the first IP!
Do you know how I can delete or modify TRYING/RINGING SDP?
Thanks
Below an example of a RINGING with SDP
# U 212.129.6.65:5060 -> 88.191.45.91:5060 SIP/2.0 180 Ringing. Allow: UPDATE,REFER. Call-ID: 866489712@192.168.95.47. Contact: sip:212.129.6.65:5060. Content-Type: application/sdp. CSeq: 21 INVITE. From: "Marc LEURENT" sip:mleurent@sd-7501.dedibox.fr;tag=542924903. Record-Route: sip:88.191.45.91;lr;ftag=542924903. Server: Cirpack/v4.39a (gw_sip). To: sip:0614730696@sd-7501.dedibox.fr;tag=01-07627-0003a50a-10152ba67. Via: SIP/2.0/UDP 88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP 192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym. Content-Length: 303. . v=0. o=cp10 118890098142 118890098144 IN IP4 212.129.46.35. s=SIP Call. c=IN IP4 212.129.47.194. t=0 0. m=audio 32896 RTP/AVP 8. b=AS:64. a=rtpmap:8 PCMA/8000/1. a=ptime:10. a=sendrecv. m=video 65534 RTP/AVP 34 31. a=rtpmap:34 H263/90000/1. a=fmtp:34 . a=rtpmap:31 H261/90000/1. a=fmtp:31 . a=inactive.
Hi Marc,
this one should be handled in the same way u do with 183/200, catch it in onreply route and force_rtpproxy()
Cheers, DanB
On 9/4/07, Marc LEURENT lftsy@free.fr wrote:
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I have a SIP provider: telecomitalia that use a cirpack! The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I don't know if it follows RFC...)
When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session description is modified. Both TRYING/RINGING with session description Connection Address/port are not replaced!
OpenWengo understand the modification and change the IP where it sends RTP from the one in TRYING/RINGING SDP to the one in 200ok (proxy address because of force_rtp_proxy()) But other softphones like Twinkle don't understand it and still send RTP to the first IP!
Do you know how I can delete or modify TRYING/RINGING SDP?
Thanks
Below an example of a RINGING with SDP
# U 212.129.6.65:5060 -> 88.191.45.91:5060 SIP/2.0 180 Ringing. Allow: UPDATE,REFER. Call-ID: 866489712@192.168.95.47. Contact: sip:212.129.6.65:5060. Content-Type: application/sdp. CSeq: 21 INVITE. From: "Marc LEURENT" sip:mleurent@sd-7501.dedibox.fr;tag=542924903. Record-Route: sip:88.191.45.91;lr;ftag=542924903. Server: Cirpack/v4.39a (gw_sip). To: sip:0614730696@sd-7501.dedibox.fr;tag=01-07627-0003a50a-10152ba67. Via: SIP/2.0/UDP 88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP 192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym. Content-Length: 303. . v=0. o=cp10 118890098142 118890098144 IN IP4 212.129.46.35. s=SIP Call. c=IN IP4 212.129.47.194. t=0 0. m=audio 32896 RTP/AVP 8. b=AS:64. a=rtpmap:8 PCMA/8000/1. a=ptime:10. a=sendrecv. m=video 65534 RTP/AVP 34 31. a=rtpmap:34 H263/90000/1. a=fmtp:34 . a=rtpmap:31 H261/90000/1. a=fmtp:31 . a=inactive. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
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Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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I already have:
onreply_route[1] { xlog("!!!! STARTING REPLY ROUTE\r\n"); #uac_restore_from(); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) { fix_nated_contact(); } exit; }
???? Strange ?? no?
Dan-Cristian Bogos a écrit :
Hi Marc,
this one should be handled in the same way u do with 183/200, catch it in onreply route and force_rtpproxy()
Cheers, DanB
On 9/4/07, Marc LEURENT lftsy@free.fr wrote: I have a SIP provider: telecomitalia that use a cirpack! The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I don't know if it follows RFC...)
When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session description is modified. Both TRYING/RINGING with session description Connection Address/port are not replaced!
OpenWengo understand the modification and change the IP where it sends RTP from the one in TRYING/RINGING SDP to the one in 200ok (proxy address because of force_rtp_proxy()) But other softphones like Twinkle don't understand it and still send RTP to the first IP!
Do you know how I can delete or modify TRYING/RINGING SDP?
Thanks
Below an example of a RINGING with SDP
# U 212.129.6.65:5060 -> 88.191.45.91:5060 SIP/2.0 180 Ringing. Allow: UPDATE,REFER. Call-ID: 866489712@192.168.95.47. Contact: sip:212.129.6.65:5060. Content-Type: application/sdp. CSeq: 21 INVITE. From: "Marc LEURENT" sip:mleurent@sd-7501.dedibox.fr;tag=542924903. Record-Route: sip:88.191.45.91;lr;ftag=542924903. Server: Cirpack/v4.39a (gw_sip). To: sip:0614730696@sd-7501.dedibox.fr;tag=01-07627-0003a50a-10152ba67. Via: SIP/2.0/UDP 88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP 192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym. Content-Length: 303. . v=0. o=cp10 118890098142 118890098144 IN IP4 212.129.46.35. s=SIP Call. c=IN IP4 212.129.47.194. t=0 0. m=audio 32896 RTP/AVP 8. b=AS:64. a=rtpmap:8 PCMA/8000/1. a=ptime:10. a=sendrecv. m=video 65534 RTP/AVP 34 31. a=rtpmap:34 H263/90000/1. a=fmtp:34 . a=rtpmap:31 H261/90000/1. a=fmtp:31 . a=inactive.
_______________________________________________ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Marc, Not at all strange. As depicted by you also, you only have it set for 183 and 200. You need a line like this: onreply_route[1] { if ((isflagset(5) || isbflagset(6)) && status=~"(100|180|183)|(2[0-9][0-9])") { ......... }
Cheers, DanB
On 9/4/07, Marc LEURENT lftsy@free.fr wrote:
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I already have:
onreply_route[1] { xlog("!!!! STARTING REPLY ROUTE\r\n"); #uac_restore_from(); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) { fix_nated_contact(); } exit;}
???? Strange ?? no?
Dan-Cristian Bogos a écrit :
Hi Marc,
this one should be handled in the same way u do with 183/200, catch it in onreply route and force_rtpproxy()
Cheers, DanB
On 9/4/07, Marc LEURENT lftsy@free.fr wrote: I have a SIP provider: telecomitalia that use a cirpack! The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I don't know if it follows RFC...)
When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session description is modified. Both TRYING/RINGING with session description Connection Address/port are not replaced!
OpenWengo understand the modification and change the IP where it sends RTP from the one in TRYING/RINGING SDP to the one in 200ok (proxy address because of force_rtp_proxy()) But other softphones like Twinkle don't understand it and still send RTP to the first IP!
Do you know how I can delete or modify TRYING/RINGING SDP?
Thanks
Below an example of a RINGING with SDP
# U 212.129.6.65:5060 -> 88.191.45.91:5060 SIP/2.0 180 Ringing. Allow: UPDATE,REFER. Call-ID: 866489712@192.168.95.47. Contact: sip:212.129.6.65:5060. Content-Type: application/sdp. CSeq: 21 INVITE. From: "Marc LEURENT" sip:mleurent@sd-7501.dedibox.fr;tag=542924903. Record-Route: sip:88.191.45.91;lr;ftag=542924903. Server: Cirpack/v4.39a (gw_sip). To: sip:0614730696@sd-7501.dedibox.fr;tag=01-07627-0003a50a-10152ba67. Via: SIP/2.0/UDP 88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP 192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym. Content-Length: 303. . v=0. o=cp10 118890098142 118890098144 IN IP4 212.129.46.35. s=SIP Call. c=IN IP4 212.129.47.194. t=0 0. m=audio 32896 RTP/AVP 8. b=AS:64. a=rtpmap:8 PCMA/8000/1. a=ptime:10. a=sendrecv. m=video 65534 RTP/AVP 34 31. a=rtpmap:34 H263/90000/1. a=fmtp:34 . a=rtpmap:31 H261/90000/1. a=fmtp:31 . a=inactive.
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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One day I will learn to think twice before saying bullshits! Thanks
Dan-Cristian Bogos a écrit :
Marc, Not at all strange. As depicted by you also, you only have it set for 183 and 200. You need a line like this: onreply_route[1] { if ((isflagset(5) || isbflagset(6)) && status=~"(100|180|183)|(2[0-9][0-9])") { ......... }
Cheers, DanB
On 9/4/07, Marc LEURENT lftsy@free.fr wrote: I already have:
onreply_route[1] { xlog("!!!! STARTING REPLY ROUTE\r\n"); #uac_restore_from(); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) { fix_nated_contact(); } exit;}
???? Strange ?? no?
Dan-Cristian Bogos a écrit :
Hi Marc,
this one should be handled in the same way u do with 183/200, catch it in onreply route and force_rtpproxy()
Cheers, DanB
On 9/4/07, Marc LEURENT lftsy@free.fr wrote: I have a SIP provider: telecomitalia that use a cirpack! The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I don't know if it follows RFC...)
When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session description is modified. Both TRYING/RINGING with session description Connection Address/port are not replaced!
OpenWengo understand the modification and change the IP where it sends RTP from the one in TRYING/RINGING SDP to the one in 200ok (proxy address because of force_rtp_proxy()) But other softphones like Twinkle don't understand it and still send RTP to the first IP!
Do you know how I can delete or modify TRYING/RINGING SDP?
Thanks
Below an example of a RINGING with SDP
# U 212.129.6.65:5060 -> 88.191.45.91:5060 SIP/2.0 180 Ringing. Allow: UPDATE,REFER. Call-ID: 866489712@192.168.95.47. Contact: sip:212.129.6.65:5060. Content-Type: application/sdp. CSeq: 21 INVITE. From: "Marc LEURENT" sip:mleurent@sd-7501.dedibox.fr;tag=542924903. Record-Route: sip:88.191.45.91;lr;ftag=542924903. Server: Cirpack/v4.39a (gw_sip). To: sip:0614730696@sd-7501.dedibox.fr;tag=01-07627-0003a50a-10152ba67. Via: SIP/2.0/UDP 88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP 192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym. Content-Length: 303. . v=0. o=cp10 118890098142 118890098144 IN IP4 212.129.46.35. s=SIP Call. c=IN IP4 212.129.47.194. t=0 0. m=audio 32896 RTP/AVP 8. b=AS:64. a=rtpmap:8 PCMA/8000/1. a=ptime:10. a=sendrecv. m=video 65534 RTP/AVP 34 31. a=rtpmap:34 H263/90000/1. a=fmtp:34 . a=rtpmap:31 H261/90000/1. a=fmtp:31 . a=inactive.
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
-
Dan-Cristian Bogos -
Marc LEURENT