Hi Sergio and List,<br><br>I'm running app_rtsp rev250 (tried rev249 also) in Asterisk is 1.6.0.10 and trying to connect to an RTSP stream provided by a GrandstreamGXV3601 IP camera. This camera works with H.264 only. Connecting to the camera using VLC RTSP client works fine (needs auth). <br>
<br>However, when trying to initiate a call both from an Eyebeam (1.5.19.5 rev build 52345) or a Linphone (3.1.2), I get the following message on the CLI :<br><br> -- Executing [554@pbx1:1] Answer("SIP/vphone-097a8bb8", "") in new stack<br>
-- Executing [554@pbx1:2] rtsp("SIP/vphone-097a8bb8", "rtsp://<a href="http://admin:admin@190.144.102.122:554">admin:admin@190.144.102.122:554</a>") in new stack<br>[091214-111022] WARNING[3599]: app_rtsp.c:1083 rtsp_play: >rtsp play<br>
[091214-111022] DEBUG[3599]: app_rtsp.c:312 GetUdpPorts: -GetUdpPorts [55617,41651]<br>[091214-111022] DEBUG[3599]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [41651,41652]<br>[091214-111022] DEBUG[3599]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [41652,41653]<br>
[091214-111022] DEBUG[3599]: app_rtsp.c:312 GetUdpPorts: -GetUdpPorts [40421,46717]<br>[091214-111022] DEBUG[3599]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [46717,46718]<br>[091214-111022] DEBUG[3599]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [46718,46719]<br>
[091214-111022] DEBUG[3599]: app_rtsp.c:451 RtspPlayerDescribe: >DESCRIBE [/]<br>[091214-111022] DEBUG[3599]: app_rtsp.c:483 RtspPlayerDescribe: <DESCRIBE [/]<br>[091214-111022] DEBUG[3599]: app_rtsp.c:1132 rtsp_play: -rtsp play loop [0]<br>
[091214-111022] DEBUG[3599]: app_rtsp.c:1211 rtsp_play: -Receiving describe<br>[091214-111022] DEBUG[3599]: app_rtsp.c:1219 rtsp_play: -Describe response code [401]<br>[091214-111022] ERROR[3599]: app_rtsp.c:1235 rtsp_play: -No Authenticate header found<br>
[091214-111022] DEBUG[3599]: app_rtsp.c:1594 rtsp_play: -rtsp_play end loop [0]<br>[091214-111022] WARNING[3599]: app_rtsp.c:1620 rtsp_play: <rtsp_play -- Executing [554@pbx1:3] Hangup("SIP/vphone-097a8bb8", "") in new stack<br>
== Spawn extension (pbx1, 554, 3) exited non-zero on 'SIP/vphone-097a8bb8'<br><br>Tried also to connect to the same RTSP flow re-streamed with VLC (which does the auth part) and then I got a:<br><br> -- Executing [553@pbx1:1] Answer("SIP/vphone-097c5100", "") in new stack <br>
-- Executing [553@pbx1:2] rtsp("SIP/vphone-097c5100", "rtsp://<a href="http://172.30.0.25:5553/test">172.30.0.25:5553/test</a>") in new stack <br>[091214-111629] WARNING[3603]: app_rtsp.c:1083 rtsp_play: >rtsp play <br>
[091214-111629] DEBUG[3603]: app_rtsp.c:312 GetUdpPorts: -GetUdpPorts [35658,41109] <br>[091214-111629] DEBUG[3603]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [41109,41110] <br>
[091214-111629] DEBUG[3603]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [41110,41111] <br>[091214-111629] DEBUG[3603]: app_rtsp.c:312 GetUdpPorts: -GetUdpPorts [54628,49715] <br>
[091214-111629] DEBUG[3603]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [49715,49716] <br>[091214-111629] DEBUG[3603]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [49716,49717] <br>
[091214-111629] DEBUG[3603]: app_rtsp.c:451 RtspPlayerDescribe: >DESCRIBE [/test] <br>[091214-111629] DEBUG[3603]: app_rtsp.c:483 RtspPlayerDescribe: <DESCRIBE [/test] <br>
[091214-111629] DEBUG[3603]: app_rtsp.c:1132 rtsp_play: -rtsp play loop [0] <br>[091214-111629] DEBUG[3603]: app_rtsp.c:1211 rtsp_play: -Receiving describe<br>[091214-111629] DEBUG[3603]: app_rtsp.c:1219 rtsp_play: -Describe response code [200]<br>
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [v=0]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [o=- 14902737644566218960 14902737644566218960 IN IP4 dexter]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [s=Unnamed]<br>
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [i=N/A]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [c=IN IP4 0.0.0.0]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [t=0 0]<br>
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=tool:vlc 1.0.3]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=recvonly]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=type:broadcast]<br>
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=charset:UTF-8]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=control:rtsp://<a href="http://172.30.0.25:5553/test">172.30.0.25:5553/test</a>]<br>
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [m=video 0 RTP/AVP 96]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:730 CreateMedia: -creating media [1,m=video 0 RTP/AVP 96]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [b=RR:0]<br>
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=rtpmap:96 H264/90000]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=fmtp:96 packetization-mode=1;profile-level-id=42e014;sprop-parameter-sets=Z0LgFNoFh8Q=,aM4wpIA=;]<br>
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=control:rtsp://<a href="http://172.30.0.25:5553/test/trackID=0">172.30.0.25:5553/test/trackID=0</a>]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [m=video 0 RTP/AVP 96]<br>
[091214-111629] DEBUG[3603]: app_rtsp.c:730 CreateMedia: -creating media [1,m=video 0 RTP/AVP 96]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [b=RR:0]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=rtpmap:96 H264/90000]<br>
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=fmtp:96 packetization-mode=1;profile-level-id=42e014;sprop-parameter-sets=Z0LgFNoFh8Q=,aM4wpIA=;]<br>[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=control:rtsp://<a href="http://172.30.0.25:5553/test/trackID=0">172.30.0.25:5553/test/trackID=0</a>]<br>
[091214-111629] DEBUG[3603]: app_rtsp.c:1330 rtsp_play: -video [2097152,96,rtsp://<a href="http://172.30.0.25:5553/test/trackID=0">172.30.0.25:5553/test/trackID=0</a>]<br>[091214-111629] ERROR[3603]: app_rtsp.c:1358 rtsp_play: No media found<br>
[091214-111629] DEBUG[3603]: app_rtsp.c:1594 rtsp_play: -rtsp_play end loop [0]<br>[091214-111629] WARNING[3603]: app_rtsp.c:1620 rtsp_play: <rtsp_play -- Executing [553@pbx1:3] Hangup("SIP/vphone-097c5100", "") in new stack<br>
== Spawn extension (pbx1, 553, 3) exited non-zero on 'SIP/vphone-097c5100'<br><br>The VLC command used (I could connect OK with several video clients to this re-streamed RTSP flow within my LAN):<br><br>vlc -vvv rtsp://<a href="http://admin:admin@192.168.0.10:554">admin:admin@192.168.0.10:554</a> --sout '#rtp{sdp=rtsp://<a href="http://0.0.0.0:5553/test">0.0.0.0:5553/test</a>}'<br>
<br>The wierd part is that loading the sample_300kbit_ulaw.3gp using VLC with Video on Demand also gives a "no media found" message. This used to work with older revisions of app_rtsp (I'm going back some revisions when there wasn't any rtsp auth implemented yet).<br>
<br><br>Relevant sip.conf:<br><br>[general]<br>language=es<br>maxexpiry=3600<br>defaultexpiry=120<br>disallow=all<br>limitonpeers=yes<br>allow=ulaw<br>allow=alaw<br>allow=gsm<br>allow=speex<br>allow=g729<br>tos_audio=ef<br>
nat=no<br>srvlookup=no<br>canreinvite=no<br>videosupport=yes<br>allow=h261<br>allow=h263<br>allow=h263p<br>allow=h264<br><br>[vphone]<br>type=friend<br>qualify=yes<br>md5secret=asdfasdfasdfasdf<br>host=dynamic<br>dtmfmode=rfc2833<br>
context=pbx1<br>callerid="vphone" <70><br>callgroup=1<br>pickupgroup=1<br>canreinvite=no<br>subscribecontext=pbx1<br>call-limit=20<br>videosupport=yes<br>allow=h261<br>allow=h263<br>allow=h263p<br>allow=h264<br>
<br>And extensions.conf:<br><br>[pbx1]<br>;Virtual PBX<br>exten => 554,1,Answer<br>exten => 554,2,rtsp(rtsp://<a href="http://admin:admin@192.168.0.10:554">admin:admin@192.168.0.10:554</a>)<br>exten => 554,3,Hangup<br>
<br>exten => 553,1,Answer<br>exten => 553,2,rtsp(rtsp://<a href="http://172.30.0.25:5553/test">172.30.0.25:5553/test</a>)<br>exten => 553,3,HangUp<br><br><br>Any suggestions on what else to test will be appreciated. I may also provide the tcpdump/wireshark capture.<br>
<br><br>Best regards,<br><br clear="all">--<br>Juan Manuel Coronado Z.<br>