Re: [SR-Users] SDPOPS issue or append_hf

I will try to look at this. It’s a bit tricky as the call-flow is not the most easiest. It’s strange because there is nothing to delete in that case because the list of codecs is already okay. Regards, Igor. De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] Envoyé : mercredi 6 août 2014 18:36 À : Igor Potjevlesch; 'Kamailio (SER) - Users Mailing List' Objet : Re: [SR-Users] SDPOPS issue or append_hf It looks related to how changes are done to a sip message. rtpproxy is working on incoming message as well as sdpops. Practically, rtpproxy adds a new line at the end of the incoming sdp. sdopos deletes from old sdp, resulting in empty lines inside the sdp. Can you do the sdpops operation before record_route() and after it call msg_apply_changes() from textopsx module? Cheers, Daniel On 06/08/14 17:48, Igor Potjevlesch wrote: It’s really linked to the initial SDP. If I have only one codec, for example G711u (plus telephone-event), and I just keep G711u, a blank line is inserted. If I keep G711u + telephone-event, everything is working fine. Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com] Envoyé : mercredi 6 août 2014 17:25 À : miconda(a)gmail.com <mailto:miconda@gmail.com> ; 'Kamailio (SER) - Users Mailing List' Objet : RE: [SR-Users] SDPOPS issue or append_hf Hello Daniel, I got a feedback from the telco in the meantime. He told me that the issue is the blank line between “rtpmap:8..” and “nortpproxy”. This parameter is supported. I have successful calls with “nortpproxy=yes”. I don’t know why sdp_keep_codecs_by_name inserts a blank line here. Regards, Igor. De : sr-users-bounces(a)lists.sip-router.org <mailto:sr-users-bounces@lists.sip-router.org> [mailto:sr-users-bounces@lists.sip-router.org] De la part de Daniel-Constantin Mierla Envoyé : mercredi 6 août 2014 16:42 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] SDPOPS issue or append_hf Hello, the problem here is with rtpproxy marker -- can you try with the parameter set to empty string? - http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp23856 Cheers, Daniel On 06/08/14 12:23, Igor Potjevlesch wrote: Hello, To be sure that the issue is not coming from append_hf, I add (…,”Call-ID”). The PAI is now inserted after the Call-ID. But, the issue remains: Content-Type: application/sdp Content-Length: 169 v=0 o=UserA 1153072414 140968390 IN IP4 A.B.C.D s=Session SDP c=IN IP4 A.B.C.D t=0 0 m=audio 60412 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=nortpproxy:yes This SDP is dropped. Someone see something missing or wrong in the SDP parts? Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com] Envoyé : mercredi 6 août 2014 11:57 À : sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> Objet : SDPOPS issue or append_hf Hello, I have an issue with the module SDPOPS while using “sdp_keep_codecs_by_name”. If the calling party sends only one codec description like: Content-Type: application/sdp Content-Length: 202 v=0 o=UserA 2966746938 1790378070 IN IP4 10.141.0.21 s=Session SDP c=IN IP4 10.141.0.21 t=0 0 m=audio 49152 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 The result of the function “sdp_keep_codecs_by_name("PCMA,PCMU,G729a");” is: Content-Type: application/sdp Content-Length: 170 P-Asserted-Identity: "+0123456789" <sip:+0123456789@sip.tld> v=0 o=UserA 2485672881 3000549892 IN IP4 a.b.c.d s=Session SDP c=IN IP4 a.b.c.d t=0 0 m=audio 40330 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=nortpproxy:yes If I open the capture in Wireshark, the PAI is not in the SDP part, and the end of the capture after “a=rtpmap:8 PCMA/8000” is seen as “Data (18 bytes)”. I don’t understand why the PAI is inserted within the SDP part. Adding the PAI is done after “sdp_keep_codecs_by_name”: if (!is_present_hf("P-Asserted-Identity")) { $var(pai) = $(fU{re.subst,/^0/+33/g}); append_hf("P-Asserted-Identity: \"$var(pai)\" <sip:$var(pai)@$fd <sip:$var%28pai%29@$fd> >\r\n"); } I guess that this cause my INVITE being dropped by 488 Media Not Acceptable Here. Regards, Igor. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA