[Asterisk-video] call handshake fails
Klaus Darilion
klaus.mailinglists at pernau.at
Wed Dec 3 12:52:07 CST 2008
Strange, for me it only works with h263p. AFAIK app_mp4 expects the
H.263 frames in h263p format.
regards
klaus
Zelalem Sintayehu wrote:
> Hi Klaus and Carlo, it is now working in Windows (X-Lite). I had to
> remove the 263+ video coved. I have now only the h263 codec. I
> think 263+ (i.e, h263-1998) may be sending bad code to asterisk (or it
> may be the client). In fact, asterisk terminates when i try to playback
> the demo file (which is encoded with h263-2000). You know, I have been
> trying different things since Monday, but after I remove the codec, it
> started working properly. But, I have still the following error message
> (uknown rtp codec 126 ...) in Asterisk.
>
> -- Executing [2000 at jain-sip:1] Answer("SIP/7503-08204e98", "") in
> new stack
> -- Executing [2000 at jain-sip:2] mp4play("SIP/7503-08204e98",
> "/home/zelalem/videos/save.mp4") in new stack
> *[Dec 4 01:16:21] NOTICE[21538]: rtp.c:1286 ast_rtp_read: Unknown RTP
> codec 126 received from '146.231.122.97'
> [Dec 4 01:16:21] NOTICE[21538]: rtp.c:1286 ast_rtp_read: Unknown RTP
> codec 126 received from '146.231.122.97'
> [Dec 4 01:16:21] NOTICE[21538]: rtp.c:1286 ast_rtp_read: Unknown RTP
> codec 126 received from '146.231.122.97'
> [Dec 4 01:16:31] NOTICE[21538]: rtp.c:1286 ast_rtp_read: Unknown RTP
> codec 126 received from '146.231.122.97'
> * -- Executing [2000 at jain-sip:3] Hangup("SIP/7503-08204e98", "") in
> new stack
> == Spawn extension (jain-sip, 2000, 3) exited non-zero on
> 'SIP/7503-08204e98'
>
> I have also seen a similar message in wireshark. I got the following
> after my send the first few (about 20) audio packets.
>
> 12633 864.046639 146.231.122.97 146.231.121.199 RTP Payload
> type=unknown (126), SSRC=2216134692, Seq=1126, Time=0
> 12634 864.046639 146.231.122.97 146.231.121.199 RTP Payload
> type=unknown (126), SSRC=2216134692, Seq=1126, Time=0
> 12635 864.046639 146.231.122.97 146.231.121.199 RTP Payload
> type=unknown (126), SSRC=2216134692, Seq=1126, Time=0
>
>
> X-Lite then send few (4) audio packets again and start sending the video
> packets. The error might have been occurred during the first video
> packets? (Intra frame problem)? I am thinking if it is a bug, you may be
> intersted to entertain it. I haven't yet tried linphone.
>
>
> - Zelalem S.
> Grahamstown, SA
>
>
>
>
> ------------------------------------------------------------------------
>
> From: zelalems at hotmail.com
> To: asterisk-video at lists.digium.com
> Date: Tue, 2 Dec 2008 12:21:41 +0300
> Subject: Re: [Asterisk-video] call handshake fails
>
> Hi Klaus and Carlo, thank you for your response. There is a little
> development. X-Lite seemed to save the video but couldn't play it back.
> You know I had to manually clicked the start button on the video pannel
> to start sending the video. Then, I have looked at the file using
> mp4info and got the following info:
> Track Type Info
> 1 audio G.711 uLaw, 5.940 secs, 64 kbps, 8000 Hz
> 2 hint Payload PCMU for track 1
> 3 video H.263, 5.844 secs, 835 kbps, 176x144 @ 26.694045 fps
> 4 hint Payload H263-1998 for track 3
>
> I couldn't play it back, though. But I have seen the file using Ubuntu's
> movie player and it is a very scrambled video. I am using logitech
> quickcam sphere mp webcam. After, I read Carlo's e-mail, then I came
> back to ubuntu and tried linphone. Again, I found out that I didn't set
> the minimum upload and download bandwidth. So, I set 135 kbit/s for
> video (I am using h263-1998 codec). Then it showed wierd characterisk
> (like terminating automatically and not starting again). I got the
> following error messge from Asterisk when I tried to record the video:
> -- Executing [7504 at jain-sip:1] Answer("SIP/7503-b7b11730", "") in
> new stack
> -- Executing [7504 at jain-sip:2] mp4save("SIP/7503-b7b11730",
> "/home/zelalem/videos/save.mp4") in new stack
> [Dec 2 18:47:54] WARNING[15692]: channel.c:2809 set_format: Unable to
> find a codec translation path from h263p to unknown
> [Dec 2 18:47:54] WARNING[15692]: app_mp4.c:804 mp4_save: mp4_save:
> Unable to set read format to ULAW|ALAW|AMRNB!
>
> And the phone terminated. I hope the above gives you an idea as to what
> my problem is. Once again, thank you. I have been doing this the past
> two weeks.
>
> Cheers,
>
> - Zelalem S.
> Grahamstown, SA
>
>
>
> > Date: Fri, 28 Nov 2008 10:40:55 +0100
> > From: klaus.mailinglists at pernau.at
> > To: asterisk-video at lists.digium.com
> > Subject: Re: [Asterisk-video] call handshake fails
> >
> > Hi!
> >
> > Thanks for the patch. Now it also works with a Sony Ericsson V800.
> >
> > regards
> > Klaus
> >
> >
> >
> > Dan Julius schrieb:
> > > Hi, Sergio,
> > >
> > > I'm following up on the problem I've been having that some (30% - 60%,
> > > not sure what it depends on) calls are not connected successfully when
> > > dialing from Samsung 3G phone to SIP client.
> > >
> > > Turns out that increasing the retransmit delay from 20 to 2000 in
> > > H324CCSRLayer::GetNextPdu seems to have resolved the problem.
> > >
> > > Are these units Milliseconds?
> > > What do you think should be a reasonable timeout?
> > >
> > > Dan
> > >
> > >
> > > On Thu, May 15, 2008 at 4:42 PM, Dan Julius <dan.julius at gmail.com
> > > <mailto:dan.julius at gmail.com>> wrote:
> > >
> > > Hi, Sergio,
> > >
> > > I actually sent these to the list a while ago, but they bounced.
> > > How do we deal with private attachments while still keeping the
> > > discussion public?
> > >
> > > Thanks for looking into this.
> > >
> > > Dan
> > >
> > > ---------- Forwarded message ----------
> > > From: *Dan Julius* <dan.julius at gmail.com <mailto:dan.julius at gmail.com>>
> > > Date: Fri, May 9, 2008 at 2:07 PM
> > > Subject: Re: [Asterisk-video] call handshake fails
> > > To: Development discussion of video media support in Asterisk
> > > <asterisk-video at lists.digium.com
> > > <mailto:asterisk-video at lists.digium.com>>
> > >
> > >
> > > Hi,
> > >
> > > Attached are logs for a call that failed. After answering the call
> > > on the mobile device, X-Lite continues to ring and nothing happens.
> > > As for video in working calls - the problem is with video from H324M
> > > to SIP. Any ideas how to debug this?
> > >
> > > Can you provide a sample for using app_transcoder?
> > >
> > > Thanks,
> > > Dan
> > >
> > >
> > >
> > > On Fri, May 9, 2008 at 1:27 PM, Sergio Garcia Murillo
> > > <sergio.garcia at fontventa.com <mailto:sergio.garcia at fontventa.com>>
> > > wrote:
> > >
> > > Could you send me a file with the h245 and h223 logs? (enable
> > > them by h324m debug level 4)
> > >
> > > The most probable cause is that you isdn provider is doing echo
> > > cancelation on the line, it usually causes random problems like
> > > this.
> > >
> > > The problem with video from SIP->H324M is that it has to be h263
> > > QCIF at maximun 52 kbs, if your videophone is not able to set
> > > this up, you'll need to use the app_transcoder module.
> > >
> > > Best regards
> > > Sergio
> > >
> > > ----- Original Message -----
> > > From: Dan Julius [mailto:dan.julius at gmail.com
> > > <mailto:dan.julius at gmail.com>]
> > > To: asterisk-video at lists.digium.com
> > > <mailto:asterisk-video at lists.digium.com>
> > > Sent: Fri, 9 May 2008 12:25:17 +0300
> > > Subject: Re: [Asterisk-video] call handshake fails
> > >
> > > Further info:
> > >
> > > - In the failed calls, the mobile phone never sends a
> > > masterSlaveDetermination packet (according to the h223 logs)
> > > - Asterisk sends the terminalCapabilitiesSet,
> > > masterSlaveDetermination and
> > > then continues to send OpenLogicalChannels.
> > >
> > > Is it OK to send OpenLogicalChannel before receiving a
> > > masterSlaveDetermination?
> > >
> > > Thanks,
> > > Dan
> > >
> > > On Fri, May 9, 2008 at 2:25 AM, Dan Julius <dan.julius at gmail.com
> > > <mailto:dan.julius at gmail.com>> wrote:
> > >
> > > > Hi, Everybody,
> > > >
> > > > I'm new to this project, so I apologize if my questions
> > > might have
> > > > already been answered elsewhere.
> > > > I am using a X-Lite, Asterisk 1.4.19, a Digium TE122 card,
> > > and a Samsung
> > > > Z720 phone.
> > > >
> > > > So far I have been able to make SIP-h234m calls (originating
> > > at either
> > > > side) with only partial success.
> > > > - I only get video in one direction, from SIP to H324M. I've
> > > read the posts
> > > > stating that SIP->H324m is actually more problematic, so I'm
> > > quite puzzled
> > > > about this.
> > > > - About 33% of the calls fail to negotiate a video
> > > connection. After
> > > > answering the call, nothing happens until I disconnect.
> > > > The out-bound h223 log of a failed call is below. Does this
> > > log indicate
> > > > that Asterisk is sending terminalCapabilitySet multiple times
> > > until it is
> > > > acknowledged?
> > > >
> > > > 1 0.000000 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2
> > > <http://2.2.2.2> H.245 terminalCapabilitySet
> > > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet
> > > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet
> > > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet
> > > > masterSlaveDetermination masterSlaveDetermination
> > > masterSlaveDetermination
> > > > masterSlaveDetermination masterSlaveDetermination
> > > masterSlaveDetermination
> > > > masterSlaveDetermination masterSlaveDetermination
> > > masterSlaveDetermination
> > > > masterSlaveDetermination masterSlaveDetermination
> > > masterSlaveDetermination
> > > > masterSlaveDetermination masterSlaveDetermination
> > > masterSlaveDetermination
> > > > 2 0.000001 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2
> > > <http://2.2.2.2> H.245 openLogicalChannel
> > > > (generic) openLogicalChannel (generic) openLogicalChannel
> > > (generic)
> > > > openLogicalChannel (generic) openLogicalChannel (generic)
> > > openLogicalChannel
> > > > (generic) openLogicalChannel (generic) openLogicalChannel
> > > (generic)
> > > > openLogicalChannel (generic) openLogicalChannel (generic)
> > > openLogicalChannel
> > > > (generic) openLogicalChannel (generic) openLogicalChannel
> > > (generic)
> > > > openLogicalChannel (h263VideoCapability) openLogicalChannel
> > > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)
> > > > openLogicalChannel (h263VideoCapability) openLogicalChannel
> > > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)
> > > > openLogicalChannel (h263VideoCapability) openLogicalChannel
> > > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)
> > > > openLogicalChannel (h263VideoCapability) openLogicalChannel
> > > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)
> > > > openLogicalChannel (h263VideoCapability) openLogicalChannel
> > > > (h263VideoCapability) multiplexEntrySend multiplexEntrySend
> > > > multiplexEntrySend multiplexEntrySend multiplexEntrySend
> > > multiplexEntrySend
> > > > multiplexEntrySend multiplexEntrySend multiplexEntrySend
> > > multiplexEntrySend
> > > > multiplexEntrySend multiplexEntrySend
> > > > 3 0.000002 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2
> > > <http://2.2.2.2> H.245 multiplexEntrySend
> > > > multiplexEntrySend multiplexEntrySend multiplexEntrySend
> > > > terminalCapabilitySetAck terminalCapabilitySetAck
> > > terminalCapabilitySetAck
> > > > terminalCapabilitySetAck terminalCapabilitySetAck
> > > terminalCapabilitySetAck
> > > > terminalCapabilitySetAck terminalCapabilitySetAck
> > > terminalCapabilitySetAck
> > > > terminalCapabilitySetAck terminalCapabilitySetAck
> > > terminalCapabilitySetAck
> > > > terminalCapabilitySetAck terminalCapabilitySetAck
> > > terminalCapabilitySetAck
> > > > terminalCapabilitySetAck
> > > > 4 0.000003 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2
> > > <http://2.2.2.2> H223
> > > > 5 0.000004 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2
> > > <http://2.2.2.2> H223
> > > >
> > > > Any pointers on how to debug this would be much appreciated.
> > > >
> > > > Thanks,
> > > > Dan
> > > >
> > > > PS - This is really great work and I'm very impressed with
> > > the project and
> > > > hope that I will be able to contribute as well.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > >
> > >
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> > >
> > >
> > >
> > >
> > >
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