Re: [Asterisk-video] Problems with app_rtsp r250 + GS GXV3601 IP Camera + H.264 Softphones

Mon, 14 Dec 2009 14:06:37 -0800

Asterisk debug logs would be of great help. Also try to get the sip negotiation to check you are sending an offer with video from linphone. A ethereal capture with the rtsp negotiation would be needed also to check the authentication part. 

Best regards
Sergio


Juan Manuel Coronado Zúñiga escribió:

Hi Sergio and List,
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).
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 :
-- Executing [...@pbx1:1] Answer("SIP/vphone-097a8bb8", "") in new stack -- Executing [...@pbx1:2] rtsp("SIP/vphone-097a8bb8", "rtsp://admin:[email protected]:554 <http://admin:[email protected]:554>") in new stack 
[091214-111022] WARNING[3599]: app_rtsp.c:1083 rtsp_play: >rtsp play
[091214-111022] DEBUG[3599]: app_rtsp.c:312 GetUdpPorts: -GetUdpPorts [55617,41651] [091214-111022] DEBUG[3599]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [41651,41652] [091214-111022] DEBUG[3599]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [41652,41653] [091214-111022] DEBUG[3599]: app_rtsp.c:312 GetUdpPorts: -GetUdpPorts [40421,46717] [091214-111022] DEBUG[3599]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [46717,46718] [091214-111022] DEBUG[3599]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [46718,46719] [091214-111022] DEBUG[3599]: app_rtsp.c:451 RtspPlayerDescribe: >DESCRIBE [/] [091214-111022] DEBUG[3599]: app_rtsp.c:483 RtspPlayerDescribe: <DESCRIBE [/] [091214-111022] DEBUG[3599]: app_rtsp.c:1132 rtsp_play: -rtsp play loop [0] [091214-111022] DEBUG[3599]: app_rtsp.c:1211 rtsp_play: -Receiving describe [091214-111022] DEBUG[3599]: app_rtsp.c:1219 rtsp_play: -Describe response code [401] [091214-111022] ERROR[3599]: app_rtsp.c:1235 rtsp_play: -No Authenticate header found [091214-111022] DEBUG[3599]: app_rtsp.c:1594 rtsp_play: -rtsp_play end loop [0] [091214-111022] WARNING[3599]: app_rtsp.c:1620 rtsp_play: <rtsp_play -- Executing [...@pbx1:3] Hangup("SIP/vphone-097a8bb8", "") in new stack == Spawn extension (pbx1, 554, 3) exited non-zero on 'SIP/vphone-097a8bb8'
Tried also to connect to the same RTSP flow re-streamed with VLC (which does the auth part) and then I got a:
-- Executing [...@pbx1:1] Answer("SIP/vphone-097c5100", "") in new stack -- Executing [...@pbx1:2] rtsp("SIP/vphone-097c5100", "rtsp://172.30.0.25:5553/test <http://172.30.0.25:5553/test>") in new stack [091214-111629] WARNING[3603]: app_rtsp.c:1083 rtsp_play: >rtsp play [091214-111629] DEBUG[3603]: app_rtsp.c:312 GetUdpPorts: -GetUdpPorts [35658,41109] [091214-111629] DEBUG[3603]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [41109,41110] [091214-111629] DEBUG[3603]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [41110,41111] [091214-111629] DEBUG[3603]: app_rtsp.c:312 GetUdpPorts: -GetUdpPorts [54628,49715] [091214-111629] DEBUG[3603]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [49715,49716] [091214-111629] DEBUG[3603]: app_rtsp.c:334 GetUdpPorts: -GetUdpPorts [49716,49717] [091214-111629] DEBUG[3603]: app_rtsp.c:451 RtspPlayerDescribe: >DESCRIBE [/test] [091214-111629] DEBUG[3603]: app_rtsp.c:483 RtspPlayerDescribe: <DESCRIBE [/test] [091214-111629] DEBUG[3603]: app_rtsp.c:1132 rtsp_play: -rtsp play loop [0] [091214-111629] DEBUG[3603]: app_rtsp.c:1211 rtsp_play: -Receiving describe [091214-111629] DEBUG[3603]: app_rtsp.c:1219 rtsp_play: -Describe response code [200] 
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [v=0]
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [o=- 14902737644566218960 14902737644566218960 IN IP4 dexter] 
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [s=Unnamed]
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [i=N/A]
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [c=IN IP4 0.0.0.0] 
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [t=0 0]
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=tool:vlc 1.0.3] 
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=recvonly]
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=type:broadcast] [091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=charset:UTF-8] [091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=control:rtsp://172.30.0.25:5553/test <http://172.30.0.25:5553/test>] [091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [m=video 0 RTP/AVP 96] [091214-111629] DEBUG[3603]: app_rtsp.c:730 CreateMedia: -creating media [1,m=video 0 RTP/AVP 96] 
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [b=RR:0]
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=rtpmap:96 H264/90000] [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=;] [091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=control:rtsp://172.30.0.25:5553/test/trackID=0 <http://172.30.0.25:5553/test/trackID=0>] [091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [m=video 0 RTP/AVP 96] [091214-111629] DEBUG[3603]: app_rtsp.c:730 CreateMedia: -creating media [1,m=video 0 RTP/AVP 96] 
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [b=RR:0]
[091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=rtpmap:96 H264/90000] [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=;] [091214-111629] DEBUG[3603]: app_rtsp.c:785 CreateSDP: -line [a=control:rtsp://172.30.0.25:5553/test/trackID=0 <http://172.30.0.25:5553/test/trackID=0>] [091214-111629] DEBUG[3603]: app_rtsp.c:1330 rtsp_play: -video [2097152,96,rtsp://172.30.0.25:5553/test/trackID=0 <http://172.30.0.25:5553/test/trackID=0>] 
[091214-111629] ERROR[3603]: app_rtsp.c:1358 rtsp_play: No media found
[091214-111629] DEBUG[3603]: app_rtsp.c:1594 rtsp_play: -rtsp_play end loop [0] [091214-111629] WARNING[3603]: app_rtsp.c:1620 rtsp_play: <rtsp_play -- Executing [...@pbx1:3] Hangup("SIP/vphone-097c5100", "") in new stack == Spawn extension (pbx1, 553, 3) exited non-zero on 'SIP/vphone-097c5100'
The VLC command used (I could connect OK with several video clients to this re-streamed RTSP flow within my LAN):
vlc -vvv rtsp://admin:[email protected]:554 <http://admin:[email protected]:554> --sout '#rtp{sdp=rtsp://0.0.0.0:5553/test <http://0.0.0.0:5553/test>}'
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). 


Relevant sip.conf:

[general]
language=es
maxexpiry=3600
defaultexpiry=120
disallow=all
limitonpeers=yes
allow=ulaw
allow=alaw
allow=gsm
allow=speex
allow=g729
tos_audio=ef
nat=no
srvlookup=no
canreinvite=no
videosupport=yes
allow=h261
allow=h263
allow=h263p
allow=h264

[vphone]
type=friend
qualify=yes
md5secret=asdfasdfasdfasdf
host=dynamic
dtmfmode=rfc2833
context=pbx1
callerid="vphone" <70>
callgroup=1
pickupgroup=1
canreinvite=no
subscribecontext=pbx1
call-limit=20
videosupport=yes
allow=h261
allow=h263
allow=h263p
allow=h264

And extensions.conf:

[pbx1]
;Virtual PBX
exten => 554,1,Answer
exten => 554,2,rtsp(rtsp://admin:[email protected]:554 <http://admin:[email protected]:554>) 
exten => 554,3,Hangup

exten => 553,1,Answer
exten => 553,2,rtsp(rtsp://172.30.0.25:5553/test <http://172.30.0.25:5553/test>) 
exten => 553,3,HangUp
Any suggestions on what else to test will be appreciated. I may also provide the tcpdump/wireshark capture. 


Best regards,

--
Juan Manuel Coronado Z.
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