Hi, Afaik, ptime difference should be perfectly legal, however, as Samuel pointed, I also heard that some implementations require symmetric ptime 20ms. Btw, in the *_default_attr(), pjmedia codec is used to set frm_ptime to native codec frame time, for G.729 it will be 10ms, and frm_per_pkt specifies the number of frames per RTP packet. Moreover, just noticed the G729 annexb difference in the SDP offer-answer, this may cause problem as they are not really compatible, please see: https://lists.cs.columbia.edu/pipermail/sip-implementors/2009-December/023988.html. BR, nanang On Fri, Mar 5, 2010 at 12:44 AM, Samuel Vinson <samuelv at laposte.net> wrote: > Hello, > > If you use my g729a port > (http://code.google.com/p/siphon/source/browse/#svn/trunk/g729a ), it seems > there is an issue with ptime : > http://code.google.com/p/siphon/issues/detail?id=374 > > In g729_default_attr function you should define : > ??? attr->info.frm_ptime = 20; // I'm not sure > ??? attr->setting.frm_per_pkt = 2; // I'm sure > > Regards > > Samuel > > Le 04/03/10 15:15, Maya Zalcberg a ?crit?: > > Hi, > I've already tried to upload it to here, so for now i will just copy the > log. > > --end msg-- > ?19:52:35.236??? pjsua_app.c? Call 0 is DISCONNECTED [reason=200 (Normal > call clearing)] > ?19:52:35.272? pjsua_media.c? Media session for call 0 is destroyed > ?19:52:36.271? pjsua_media.c? Closing sound device after idle for 1 seconds > ?19:52:36.271? pjsua_media.c? Closing iPhone Sound Device sound playback > device and iPhone Sound Device sound capture device > ?19:52:59.091?? pjsua_core.c? RX 823 bytes Request msg INVITE/cseq=225308893 > (rdata0x84d064) from UDP 80.179.0.4:5060: > INVITE sip:722511874 at 192.168.1.59:5060 SIP/2.0 > Via: SIP/2.0/UDP 80.179.0.4:5060;branch=z9hG4bKgqo3mo00d0b10koup7c0.1 > From: "a > 1"<sip:1867 at 072.012.net;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1"<sip:722511874 at 072.012.net> > Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 > CSeq: 225308893 INVITE > Contact: <sip:1867 at 80.179.0.4:5060;transport=udp> > Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY > Accept: multipart/mixed,application/media_control+xml,application/sdp > Supported: > Max-Forwards: 4 > Content-Type: application/sdp > Content-Length: 227 > > v=0 > o=BroadWorks 162350108 1 IN IP4 212.199.138.8 > s=- > c=IN IP4 212.199.138.8 > t=0 0 > m=audio 21758 RTP/AVP 18 101 > a=fmtp:18 annexb=yes > a=fmtp:101 0-15 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > a=sendrecv > > --end msg-- > ?19:52:59.092? pjsua_media.c? Media index 0 selected for call 1 > ?19:52:59.094?? pjsua_core.c? TX 344 bytes Response msg > 100/INVITE/cseq=225308893 (tdta0x869c00) to UDP 80.179.0.4:5060: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 80.179.0.4:5060;received=80.179.0.4;branch=z9hG4bKgqo3mo00d0b10koup7c0.1 > Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 > From: "a 1" > <sip:1867 at 072.012.net;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1" <sip:722511874 at 072.012.net> > CSeq: 225308893 INVITE > Content-Length:? 0 > > > --end msg-- > ?19:52:59.104? pjsua_media.c? Opening sound device PCM at 16000/1/20ms > ?19:52:59.235???? ec0x24d100? Echo suppressor created, clock_rate=16000, > channel=1, samples per frame=320, tail length=200 ms, latency=140 ms > ?19:52:59.459?? conference.c? Port 2 (ring) transmitting to port 0 (iPhone > Sound Device) > ?19:52:59.468??? pjsua_app.c? Incoming call for account 2! > From: "a 1" <sip:1867@xxxxxxxxxxx;user=phone> > To: "a 1" <sip:722511874 at 072.012.net> > Press a to answer or h to reject call > 2010-03-01 19:52:59.498 MySIPhoneV2[1527:207] Could not load the "siax.png" > image referenced from a nib in the bundle with identifier > "com.zemingo.MySIPhoneV2" > ?19:53:00.564?? strm0x871b74? VAD temporarily disabled > ?19:53:00.565?? strm0x871b74? Encoder stream started > ?19:53:00.565?? strm0x871b74? Decoder stream started > ?19:53:00.565? pjsua_media.c? Media updates, stream #0: G729 (sendrecv) > ?19:53:00.566?? conference.c? Port 2 (ring) stop transmitting to port 0 > (iPhone Sound Device) > ?19:53:00.566?? conference.c? Port 3 (sip:1867 at 072.012.net;user=phone) > transmitting to port 0 (iPhone Sound Device) > ?19:53:00.566?? conference.c? Port 0 (iPhone Sound Device) transmitting to > port 3 (sip:1867 at 072.012.net;user=phone) > ?19:53:00.566??? pjsua_app.c? Media for call 1 is active > ?19:53:00.566?? pjsua_core.c? TX 869 bytes Response msg > 200/INVITE/cseq=225308893 (tdta0x869c00) to UDP 80.179.0.4:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 80.179.0.4:5060;received=80.179.0.4;branch=z9hG4bKgqo3mo00d0b10koup7c0.1 > Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 > From: "a 1" > <sip:1867 at 072.012.net;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1" <sip:722511874 at 072.012.net>;tag=xbC9.pmCU9.-vEm7FhtxB25YcK.RmCye > CSeq: 225308893 INVITE > Contact: <sip:722511874 at 192.168.1.59:5060> > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, > MESSAGE, OPTIONS > Supported: replaces, 100rel, timer, norefersub > Content-Type: application/sdp > Content-Length:?? 274 > > v=0 > o=- 3476454779 3476454780 IN IP4 192.168.1.59 > s=pjmedia > c=IN IP4 192.168.1.59 > t=0 0 > a=X-nat:0 > m=audio 4002 RTP/AVP 18 101 > a=rtcp:4003 IN IP4 192.168.1.59 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > --end msg-- > ?19:53:00.569??? pjsua_app.c? Call 1 state changed to CONNECTING > ?19:53:01.096?? pjsua_core.c? TX 869 bytes Response msg > 200/INVITE/cseq=225308893 (tdta0x869c00) to UDP 80.179.0.4:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 80.179.0.4:5060;received=80.179.0.4;branch=z9hG4bKgqo3mo00d0b10koup7c0.1 > Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 > From: "a 1" > <sip:1867 at 072.012.net;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1" <sip:722511874 at 072.012.net>;tag=xbC9.pmCU9.-vEm7FhtxB25YcK.RmCye > CSeq: 225308893 INVITE > Contact: <sip:722511874 at 192.168.1.59:5060> > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, > MESSAGE, OPTIONS > Supported: replaces, 100rel, timer, norefersub > Content-Type: application/sdp > Content-Length:?? 274 > > v=0 > o=- 3476454779 3476454780 IN IP4 192.168.1.59 > s=pjmedia > c=IN IP4 192.168.1.59 > t=0 0 > a=X-nat:0 > m=audio 4002 RTP/AVP 18 101 > a=rtcp:4003 IN IP4 192.168.1.59 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > --end msg-- > ?19:53:02.111?? pjsua_core.c? TX 869 bytes Response msg > 200/INVITE/cseq=225308893 (tdta0x869c00) to UDP 80.179.0.4:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 80.179.0.4:5060;received=80.179.0.4;branch=z9hG4bKgqo3mo00d0b10koup7c0.1 > Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 > From: "a 1" > <sip:1867 at 072.012.net;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1" <sip:722511874 at 072.012.net>;tag=xbC9.pmCU9.-vEm7FhtxB25YcK.RmCye > CSeq: 225308893 INVITE > Contact: <sip:722511874 at 192.168.1.59:5060> > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, > MESSAGE, OPTIONS > Supported: replaces, 100rel, timer, norefersub > Content-Type: application/sdp > Content-Length:?? 274 > > v=0 > o=- 3476454779 3476454780 IN IP4 192.168.1.59 > s=pjmedia > c=IN IP4 192.168.1.59 > t=0 0 > a=X-nat:0 > m=audio 4002 RTP/AVP 18 101 > a=rtcp:4003 IN IP4 192.168.1.59 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > --end msg-- > ?19:53:02.211?? pjsua_core.c? RX 450 bytes Request msg ACK/cseq=225308893 > (rdata0x84d064) from UDP 80.179.0.4:5060: > ACK sip:722511874 at 192.168.1.59:5060 SIP/2.0 > Via: SIP/2.0/UDP 80.179.0.4:5060;branch=z9hG4bK7d5nfa1060s0nmg3r3k0.1 > From: "a 1" > <sip:1867 at 072.012.net;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1" <sip:722511874 at 072.012.net>;tag=xbC9.pmCU9.-vEm7FhtxB25YcK.RmCye > Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 > CSeq: 225308893 ACK > Contact: <sip:1867 at 80.179.0.4:5060;transport=udp> > Max-Forwards: 4 > Content-Length: 0 > > > --end msg-- > ?19:53:02.214??? pjsua_app.c? Call 1 state changed to CONFIRMED > > > Regards, > Maya > > > > > _______________________________________________ > 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 > > > _______________________________________________ > 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 > >
