[Asterisk-video] Problems with app_rtsp r250 + GS GXV3601 IP Camera + H.264 Softphones
Sergio Garcia Murillo
sergio.garcia at fontventa.com
Mon Dec 14 16:03:15 CST 2009
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 [554 at pbx1:1] Answer("SIP/vphone-097a8bb8", "") in new
> stack
> -- Executing [554 at pbx1:2] rtsp("SIP/vphone-097a8bb8",
> "rtsp://admin:admin@190.144.102.122:554
> <http://admin:admin@190.144.102.122:554>") 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 [554 at 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 [553 at pbx1:1] Answer("SIP/vphone-097c5100", "") in new
> stack
> -- Executing [553 at 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 [553 at 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:admin@192.168.0.10:554
> <http://admin:admin@192.168.0.10:554> --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:admin@192.168.0.10:554
> <http://admin:admin@192.168.0.10:554>)
> 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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