[Asterisk-video] call handshake fails

Wed Dec 3 10:14:39 CST 2008

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=012635 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.comTo: asterisk-video at lists.digium.comDate: Tue, 2 Dec 2008 12:21:41 +0300Subject: 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    Info1    audio    G.711 uLaw, 5.940 secs, 64 kbps, 8000 Hz2    hint    Payload PCMU for track 13    video    H.263, 5.844 secs, 835 kbps, 176x144 @ 26.694045 fps4    hint    Payload H263-1998 for track 3I 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.> > >> > >> > >> > >> > >> > >> > >> > > > > > _______________________________________________> > --Bandwidth and Colocation Provided by http://www.api-digital.com--> > > > asterisk-video mailing list> > To UNSUBSCRIBE or update options visit:> > http://lists.digium.com/mailman/listinfo/asterisk-video> > > > > > > > > > > > ------------------------------------------------------------------------> > > > _______________________________________________> > --Bandwidth and Colocation Provided by http://www.api-digital.com--> > > > asterisk-video mailing list> > To UNSUBSCRIBE or update options visit:> > http://lists.digium.com/mailman/listinfo/asterisk-video> > _______________________________________________> --Bandwidth and Colocation Provided by http://www.api-digital.com--> > asterisk-video mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-video

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