Hi Sergio and list,<br><br>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.<br>
<br>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.<br><br>I'm sending the Asterisk debug logs and the tcpdump capture off the list.<br>
<br>Thank you and best regards,<br><br><br clear="all">--<br>Juan Manuel Coronado Z.<br>
<br><br><div class="gmail_quote">On Mon, Dec 14, 2009 at 5:03 PM, Sergio Garcia Murillo <span dir="ltr"><<a href="mailto:sergio.garcia@fontventa.com">sergio.garcia@fontventa.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000">
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.<br>
A ethereal capture with the rtsp negotiation would be needed also to
check the authentication part.<br>
<br>
Best regards<br>
Sergio<br>
<br>
<br>
Juan Manuel Coronado Zúñiga escribió:
<blockquote type="cite"><div><div></div><div class="h5">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" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">admin:admin@192.168.0.10:554</a>
--sout '#rtp{sdp=rtsp://<a href="http://0.0.0.0:5553/test" target="_blank">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" target="_blank">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" target="_blank">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>
</div></div><pre><hr size="4" width="90%">
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</blockquote>
<br>
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