Re: [SR-Users] SDPOPS issue or append_hf

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; '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@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