I use one phone to call my sip client, then I answer it and negotiate the media information, but one media stream needs to be updated, and I got the assertion `neg && !pjmedia_sdp_neg_was_answer_remote(neg)'. The log is like this: 04:28:40.502 os_core_unix.c !pjlib 2.0.1 for POSIX initialized 04:28:40.505 sip_endpoint.c .Creating endpoint instance... 04:28:40.505 pjlib .select() I/O Queue created (0xb7c900) 04:28:40.506 sip_endpoint.c .Module "mod-msg-print" registered 04:28:40.506 sip_transport. .Transport manager created. 04:28:40.506 pjsua_core.c .PJSUA state changed: NULL --> CREATED 04:28:40.506 sip_endpoint.c .Module "mod-pjsua-log" registered 04:28:40.506 sip_endpoint.c .Module "mod-tsx-layer" registered 04:28:40.506 sip_endpoint.c .Module "mod-stateful-util" registered 04:28:40.506 sip_endpoint.c .Module "mod-ua" registered 04:28:40.507 sip_endpoint.c .Module "mod-100rel" registered 04:28:40.507 sip_endpoint.c .Module "mod-pjsua" registered 04:28:40.507 sip_endpoint.c .Module "mod-invite" registered 04:28:40.509 pa_dev.c ..PortAudio sound library initialized, status=0 04:28:40.509 pa_dev.c ..PortAudio host api count=1 04:28:40.509 pa_dev.c ..Sound device count=0 04:28:40.510 pjlib ..select() I/O Queue created (0xb89288) 04:28:40.533 sip_endpoint.c .Module "mod-evsub" registered 04:28:40.533 sip_endpoint.c .Module "mod-presence" registered 04:28:40.533 sip_endpoint.c .Module "mod-mwi" registered 04:28:40.534 sip_endpoint.c .Module "mod-refer" registered 04:28:40.534 sip_endpoint.c .Module "mod-pjsua-pres" registered 04:28:40.534 sip_endpoint.c .Module "mod-pjsua-im" registered 04:28:40.534 sip_endpoint.c .Module "mod-pjsua-options" registered 04:28:40.534 pjsua_core.c .1 SIP worker threads created 04:28:40.535 pjsua_core.c .pjsua version 2.0.1 for Linux-3.2.0.29/x86_64/glibc-2.15 initialized 04:28:40.535 pjsua_core.c .PJSUA state changed: CREATED --> INIT 04:28:40.536 tcplis:5060 SIP TCP listener ready for incoming connections at 192.168.96.63:5060 04:28:40.536 pjsua_core.c PJSUA state changed: INIT --> STARTING 04:28:40.536 pjsua_media.c ..NAT type detection failed: Invalid STUN server or server not configured (PJNATH_ESTUNINSERVER) 04:28:40.536 sip_endpoint.c .Module "mod-unsolicited-mwi" registered 04:28:40.536 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING 04:28:40.537 pjsua_acc.c Adding account: id=sip:4444 at 192.168.96.63 04:28:40.537 pjsua_acc.c .Account sip:4444 at 192.168.96.63 added with id 0 04:28:40.537 pjsua_acc.c .Acc 0: setting registration.. 04:28:40.537 pjsua_acc.c ..Contact for acc 0 updated for SIP outbound: <sip:4444 at 192.168.96.63:5060;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000a3fe1a18>" 04:28:40.537 tcpc0xba1648 ...TCP client transport created 04:28:40.545 tcpc0xba1648 ...TCP transport 192.168.96.63:55291 is connecting to 192.168.96.63:5060... 04:28:40.545 pjsua_core.c ...TX 622 bytes Request msg REGISTER/cseq=36082 (tdta0xb9ee50) to tcp 192.168.96.63:5060: REGISTER sip:192.168.96.63;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 192.168.96.63:55291;rport;branch=z9hG4bKPjeac63969-4789-4366-af55-4cecf1a05a7e Max-Forwards: 70 From: <sip:4444@192.168.96.63>;tag=d8c8abaa-78b3-4466-9e75-d60a01ce37c9 To: <sip:4444 at 192.168.96.63> Call-ID: 304e94c2-f655-49c5-a313-dcdfa7cbb50f CSeq: 36082 REGISTER Supported: outbound, path Contact: <sip:4444 at 192.168.96.63:5060;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000a3fe1a18>" Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 --end msg-- 04:28:40.545 pjsua_acc.c ..Acc 0: Registration sent 04:28:40.545 pjsua_aud.c Setting null sound device.. 04:28:40.545 pjsua_aud.c .Opening null sound device.. 04:28:40.545 tcplis:5060 !TCP listener 192.168.96.63:5060: got incoming TCP connection from 192.168.96.63:55291, sock=8 04:28:40.553 tcps0x7fdf3000 TCP server transport created 04:28:40.553 tcpc0xba1648 TCP transport 192.168.96.63:55291 is connected to 192.168.96.63:5060 04:28:40.554 pjsua_core.c .RX 622 bytes Request msg REGISTER/cseq=36082 (rdata0x7fdf30000df8) from tcp 192.168.96.63:55291: REGISTER sip:192.168.96.63;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 192.168.96.63:55291;rport;branch=z9hG4bKPjeac63969-4789-4366-af55-4cecf1a05a7e Max-Forwards: 70 From: <sip:4444@192.168.96.63>;tag=d8c8abaa-78b3-4466-9e75-d60a01ce37c9 To: <sip:4444 at 192.168.96.63> Call-ID: 304e94c2-f655-49c5-a313-dcdfa7cbb50f CSeq: 36082 REGISTER Supported: outbound, path Contact: <sip:4444 at 192.168.96.63:5060;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000a3fe1a18>" Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 --end msg-- 04:28:40.571 sip_endpoint.c .Message Request msg REGISTER/cseq=36082 (rdata0x7fdf30000df8) from 192.168.96.63:55291 was dropped/unhandled by any modules 04:28:40.817 tcplis:5060 TCP listener 192.168.96.63:5060: got incoming TCP connection from 192.168.32.58:47501, sock=9 04:28:40.817 tcps0x7fdf3000 TCP server transport created 04:28:40.817 pjsua_core.c .RX 996 bytes Request msg INVITE/cseq=101 (rdata0x7fdf30004c58) from tcp 192.168.32.58:47501: INVITE sip:4444 at 192.168.96.63:5060 SIP/2.0 Via: SIP/2.0/TCP 192.168.32.58:5060;branch=z9hG4bK1e716c9a2863 From: <sip:6005@192.168.32.58>;tag=fadb329c-03ed-490a-9aed-4af004a5ddc3-20469019 To: <sip:4444 at 192.168.96.63> Date: Wed, 15 Aug 2012 11:28:41 GMT Call-ID: 5c07c000-2b187e9-283-3a20a8c0 at 192.168.32.58 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM8.0 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Contact: <sip:6005 at 192.168.32.58:5060;transport=tcp> Expires: 180 Allow-Events: presence, kpml Supported: X-cisco-srtp-fallback Supported: Geolocation Call-Info: <sip:192.168.32.58:5060>;method="NOTIFY;Event=telephone-event;Duration=500" Cisco-Guid: 1544011776-0000065536-0000000536-0975218880 Session-Expires: 1800 P-Asserted-Identity: <sip:6005 at 192.168.32.58> Remote-Party-ID: <sip:6005 at 192.168.32.58>;party=calling;screen=yes;privacy=off Max-Forwards: 70 Content-Length: 0 --end msg-- 04:28:40.817 pjsua_call.c .Incoming Request msg INVITE/cseq=101 (rdata0x7fdf30004c58) 04:28:40.817 pjsua_core.c .....TX 313 bytes Response msg 100/INVITE/cseq=101 (tdta0x7fdf3000a7e0) to tcp 192.168.32.58:47501: SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.32.58:5060;received=192.168.32.58;branch=z9hG4bK1e716c9a2863 Call-ID: 5c07c000-2b187e9-283-3a20a8c0 at 192.168.32.58 From: <sip:6005@192.168.32.58>;tag=fadb329c-03ed-490a-9aed-4af004a5ddc3-20469019 To: <sip:4444 at 192.168.96.63> CSeq: 101 INVITE Content-Length: 0 --end msg-- 04:28:40.817 APP ..Incoming call from <sip:6005 at 192.168.32.58>!! 04:28:40.817 pjsua_call.c ..Answering call 0: code=200 04:28:40.817 pjsua_media.c ...Call 0: initializing media.. 04:28:40.818 pjsua_media.c ....RTP socket reachable at 192.168.96.63:40000 04:28:40.818 pjsua_media.c ....RTCP socket reachable at 192.168.96.63:40001 04:28:40.818 pjsua_media.c ....Media index 0 selected for audio call 0 04:28:40.818 pjsua_core.c ......TX 1120 bytes Response msg 200/INVITE/cseq=101 (tdta0x7fdf3000a7e0) to tcp 192.168.32.58:47501: SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.32.58:5060;received=192.168.32.58;branch=z9hG4bK1e716c9a2863 Call-ID: 5c07c000-2b187e9-283-3a20a8c0 at 192.168.32.58 From: <sip:6005@192.168.32.58>;tag=fadb329c-03ed-490a-9aed-4af004a5ddc3-20469019 To: <sip:4444 at 192.168.96.63>;tag=c3e37297-d6d5-4959-b88e-7117f2eaee84 CSeq: 101 INVITE Contact: <sip:4444 at 192.168.96.63:5060;transport=TCP;ob> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Length: 502 v=0 o=- 3554018920 3554018920 IN IP4 192.168.96.63 s=pjmedia c=IN IP4 192.168.96.63 b=AS:84 t=0 0 a=X-nat:0 m=audio 40000 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 192.168.96.63 b=TIAS:64000 a=rtcp:40001 IN IP4 192.168.96.63 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 --end msg-- 04:28:40.818 APP .........Call 0 state=CONNECTING 04:28:40.879 pjsua_core.c .RX 695 bytes Request msg ACK/cseq=101 (rdata0x7fdf30004c58) from tcp 192.168.32.58:47501: ACK sip:4444 at 192.168.96.63:5060;transport=TCP;ob SIP/2.0 Via: SIP/2.0/TCP 192.168.32.58:5060;branch=z9hG4bK1e7247529c79 From: <sip:6005@192.168.32.58>;tag=fadb329c-03ed-490a-9aed-4af004a5ddc3-20469019 To: <sip:4444 at 192.168.96.63>;tag=c3e37297-d6d5-4959-b88e-7117f2eaee84 Date: Wed, 15 Aug 2012 11:28:41 GMT Call-ID: 5c07c000-2b187e9-283-3a20a8c0 at 192.168.32.58 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: presence, kpml Content-Type: application/sdp Content-Length: 212 v=0 o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.32.58 s=SIP Call c=IN IP4 192.168.98.142 t=0 0 m=audio 24650 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 --end msg-- 04:28:40.879 pjsua_media.c ...Call 0: updating media.. 04:28:40.879 pjsua_aud.c ....Audio channel update.. 04:28:40.880 strm0x7fdf3001 .....VAD temporarily disabled 04:28:40.880 strm0x7fdf3001 .....Encoder stream started 04:28:40.880 strm0x7fdf3001 .....Decoder stream started 04:28:40.880 pjsua_media.c ....Audio updated, stream #0: PCMU (sendrecv) vgw: /home/eccom/videogateway/pjproject-2.0.1/pjsip/src/pjsua-lib/pjsua_media.c:2522: pjsua_media_channel_update: Assertion `neg && !pjmedia_sdp_neg_was_answer_remote(neg)' failed. 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