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 > <http://80.179.0.4:5060>: > INVITE sip:722511874 at 192.168.1.59:5060 > <http://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@xxxxxxxxxxx > <mailto:sip%3A1867 at 072.012.net>;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1"<sip:722511874 at 072.012.net <mailto:sip%3A722511874 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 > <http://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@xxxxxxxxxxx > <mailto:sip%3A1867 at 072.012.net>;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1" <sip:722511874 at 072.012.net <mailto:sip%3A722511874 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 > <mailto:sip%3A1867 at 072.012.net>;user=phone> > To: "a 1" <sip:722511874 at 072.012.net <mailto:sip%3A722511874 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 > <mailto:sip%3A1867 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 > <mailto:sip%3A1867 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 > <http://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@xxxxxxxxxxx > <mailto:sip%3A1867 at 072.012.net>;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1" <sip:722511874 at 072.012.net > <mailto:sip%3A722511874 at 072.012.net>>;tag=xbC9.pmCU9.-vEm7FhtxB25YcK.RmCye > CSeq: 225308893 INVITE > Contact: <sip:722511874 at 192.168.1.59:5060 > <http://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 > <http://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@xxxxxxxxxxx > <mailto:sip%3A1867 at 072.012.net>;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1" <sip:722511874 at 072.012.net > <mailto:sip%3A722511874 at 072.012.net>>;tag=xbC9.pmCU9.-vEm7FhtxB25YcK.RmCye > CSeq: 225308893 INVITE > Contact: <sip:722511874 at 192.168.1.59:5060 > <http://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 > <http://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@xxxxxxxxxxx > <mailto:sip%3A1867 at 072.012.net>;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1" <sip:722511874 at 072.012.net > <mailto:sip%3A722511874 at 072.012.net>>;tag=xbC9.pmCU9.-vEm7FhtxB25YcK.RmCye > CSeq: 225308893 INVITE > Contact: <sip:722511874 at 192.168.1.59:5060 > <http://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 > <http://80.179.0.4:5060>: > ACK sip:722511874 at 192.168.1.59:5060 > <http://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@xxxxxxxxxxx > <mailto:sip%3A1867 at 072.012.net>;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- > To: "a 1" <sip:722511874 at 072.012.net > <mailto:sip%3A722511874 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 > -------------- next part -------------- An HTML attachment was scrubbed... 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