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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
cite="mid:c8d9da6f0912141332g410beaffg20530360d996948d@mail.gmail.com"
type="cite">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
moz-do-not-send="true" href="http://admin:admin@190.144.102.122:554">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
moz-do-not-send="true" href="http://172.30.0.25:5553/test">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 moz-do-not-send="true"
href="http://172.30.0.25:5553/test">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 moz-do-not-send="true"
href="http://172.30.0.25:5553/test/trackID=0">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 moz-do-not-send="true"
href="http://172.30.0.25:5553/test/trackID=0">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 moz-do-not-send="true"
href="http://172.30.0.25:5553/test/trackID=0">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 moz-do-not-send="true"
href="http://admin:admin@192.168.0.10:554">admin:admin@192.168.0.10:554</a>
--sout '#rtp{sdp=rtsp://<a moz-do-not-send="true"
href="http://0.0.0.0:5553/test">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 moz-do-not-send="true"
href="http://admin:admin@192.168.0.10:554">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 moz-do-not-send="true"
href="http://172.30.0.25:5553/test">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>
<pre wrap="">
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</blockquote>
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