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

Thu Dec 17 15:09:23 CST 2009

Hi Sergio and list,

Thank you for answering. I must say that forcing the SIP negotiation to send
video before attempting the call worked with the Eyebeam (other PC with
version 1.5.7 build 31159) to the re-streamed RTSP flow with VLC (i.e.
without the authentication). With the Linphone, the call is stablished but
no video is shown. Calling to the RTSP that connects directly to the camera
also fails the authentication.

The bad news are that when the succesfull call is terminated, Asterisk
crashes. I'll try with other versions of Asterisk 1.6.x (currently using
1.6.0.10) to see how it goes.

I'm sending the Asterisk debug logs and the tcpdump capture off the list.

Thank you and best regards,

--
Juan Manuel Coronado Z.

On Mon, Dec 14, 2009 at 5:03 PM, Sergio Garcia Murillo <
sergio.garcia at fontventa.com> wrote:

>  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 at 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") 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]
> [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]
> [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]
> [091214-111629] DEBUG[3603]: app_rtsp.c:1330 rtsp_play: -video
> [2097152,96,rtsp://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 --sout '#rtp{sdp=rtsp://
> 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)
> exten => 554,3,Hangup
>
> exten => 553,1,Answer
> exten => 553,2,rtsp(rtsp://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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