On 1/11/08, Alain Totouom <alain.totouom at gmx.de> wrote: > > Hi, > > while debugging a mis-configured sip-phone we ran into this strange > behavior of > the stack. > > The phone sends an INVITE with two medias in its offer (audio & video): > m=audio 51788 RTP/AVP 0 3 8 101 > m=video 57688 RTP/AVP 34 > > Instead of sending a 200 OK with a valid media description for the audio > and a > mute or something like that for the video it does yet not supports, the > stack > just send a 415 with no information about the failure that has occurred as > mandated by rfc.3261 (21.4.13 page 186). > Especially none of the mandatory headers (Accept, Accept-Encoding, or > Accept-Language) to be used in such a case are used. > > Our quick hack was to add the Accept Header rfc.2616 (?14.1) > > Accept: audio/* > > in the response and the phone did show up afterwards with only one media > in its > offer this time ;o) > > > But wouldn't it be better to just accept the call since PJSIP can handle > the > audio stream ;o) and just set the port for the unsupported media to 0? This is supposed to be the case (well, at least with recent pjsip version). Which version do you use? cheers, -benny Cheers > Alain > > > > > > > INVITE sip:100001 at 192.168.001.100:5060 SIP/2.0 > Record-Route: <sip:123.123.123.123;lr=on;ftag=as09234324> > Record-Route: <sip:234.234.234.234;lr=on> > Record-Route: <sip:123.123.123.123;lr=on;ftag=as09234324> > Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4b2222.23456.9 > Via: SIP/2.0/UDP 234.234.234.234;branch=z9hG4b2222.23456.9 > Via: SIP/2.0/UDP 123.123.123.123:5060;received=217.10.68.226 > ;branch=z9hG4b45c12345 > Via: SIP/2.0/UDP 44.44.44.44:5060;branch=z9hG4b45c12345;rport=5060 > From: "xyzFon" <sip:2000002@xxxxxxxxxx>;tag=as09234324 > To: <sip:1234567 at foobar.xxx> > Contact: <sip:2000002 at 44.44.44.44:5060> > Call-ID: 2a6d8ddf684bf7718dd210f734eee at foobar.xxx > CSeq: 103 INVITE > Max-Forwards: 67 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 311 > > v=0 > o=root 19999 20001 IN IP4 123.123.123.123 > s=session > c=IN IP4 123.123.123.123 > t=0 0 > m=audio 51788 RTP/AVP 0 3 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > m=video 57688 RTP/AVP 34 > a=rtpmap:34 H263/90000 > > > > SIP/2.0 415 Unsupported Media Type > Via: SIP/2.0/UDP > 123.123.123.123:5060;received=123.123.123.123;branch=z9hG4b2222.23456.9 > Via: SIP/2.0/UDP 234.234.234.234;branch=z9hG4b2222.23456.9 > Via: SIP/2.0/UDP 123.123.123.123:5060;received=217.10.68.226 > ;branch=z9hG4b45c12345 > Via: SIP/2.0/UDP 44.44.44.44:5060;rport=5060;branch=z9hG4b45c12345 > Record-Route: <sip:123.123.123.123;lr;ftag=as09234324> > Record-Route: <sip:234.234.234.234;lr> > Record-Route: <sip:123.123.123.123;lr;ftag=as09234324> > Call-ID: 2a6d8ddf684bf7718dd210f734eee at foobar.xxx > From: "xyzFon" <sip:2000002@xxxxxxxxxx>;tag=as09234324 > To: <sip:1234567 at foobar.xxx>;tag=8ca73284cf704cff8d6f09a50e520f8b > CSeq: 103 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > PUBLISH, > REFER, MESSAGE, OPTIONS > Supported: replaces, 100rel, norefersub > Content-Length: 0 > > > > -- > "" > (o)(o) > _____o00o__(__)__o00o_____ > 1024D/A9F85A52 2000-01-18 Alain Totouom <totouom at gmx.de> > PGP FingerPrint DA180DF2 FBD25F67 0656452D E3A27531 A9F85A52 > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080111/42df69d5/attachment-0001.html
