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

Mon Dec 14 15:32:06 CST 2009

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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