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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:<br> Track Type Info<br>1 audio G.711 uLaw, 5.940 secs, 64 kbps, 8000 Hz<br>2 hint Payload PCMU for track 1<br>3 video H.263, 5.844 secs, 835 kbps, 176x144 @ 26.694045 fps<br>4 hint Payload H263-1998 for track 3<br><br>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:<br> -- Executing [7504@jain-sip:1] Answer("SIP/7503-b7b11730", "") in new stack<br> -- Executing [7504@jain-sip:2] mp4save("SIP/7503-b7b11730", "/home/zelalem/videos/save.mp4") in new stack<br>[Dec 2 18:47:54] WARNING[15692]: channel.c:2809 set_format: Unable to find a codec translation path from h263p to unknown<br>[Dec 2 18:47:54] WARNING[15692]: app_mp4.c:804 mp4_save: mp4_save: Unable to set read format to ULAW|ALAW|AMRNB!<br><br>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.<br><br>Cheers,<br><br>- Zelalem S. <br>Grahamstown, SA<br><br><br><br>> Date: Fri, 28 Nov 2008 10:40:55 +0100<br>> From: klaus.mailinglists@pernau.at<br>> To: asterisk-video@lists.digium.com<br>> Subject: Re: [Asterisk-video] call handshake fails<br>> <br>> Hi!<br>> <br>> Thanks for the patch. Now it also works with a Sony Ericsson V800.<br>> <br>> regards<br>> Klaus<br>> <br>> <br>> <br>> Dan Julius schrieb:<br>> > Hi, Sergio,<br>> > <br>> > I'm following up on the problem I've been having that some (30% - 60%, <br>> > not sure what it depends on) calls are not connected successfully when <br>> > dialing from Samsung 3G phone to SIP client.<br>> > <br>> > Turns out that increasing the retransmit delay from 20 to 2000 in <br>> > H324CCSRLayer::GetNextPdu seems to have resolved the problem.<br>> > <br>> > Are these units Milliseconds?<br>> > What do you think should be a reasonable timeout?<br>> > <br>> > Dan<br>> > <br>> > <br>> > On Thu, May 15, 2008 at 4:42 PM, Dan Julius <dan.julius@gmail.com <br>> > <mailto:dan.julius@gmail.com>> wrote:<br>> > <br>> > Hi, Sergio,<br>> > <br>> > I actually sent these to the list a while ago, but they bounced.<br>> > How do we deal with private attachments while still keeping the<br>> > discussion public?<br>> > <br>> > Thanks for looking into this.<br>> > <br>> > Dan<br>> > <br>> > ---------- Forwarded message ----------<br>> > From: *Dan Julius* <dan.julius@gmail.com <mailto:dan.julius@gmail.com>><br>> > Date: Fri, May 9, 2008 at 2:07 PM<br>> > Subject: Re: [Asterisk-video] call handshake fails<br>> > To: Development discussion of video media support in Asterisk<br>> > <asterisk-video@lists.digium.com<br>> > <mailto:asterisk-video@lists.digium.com>><br>> > <br>> > <br>> > Hi,<br>> > <br>> > Attached are logs for a call that failed. After answering the call<br>> > on the mobile device, X-Lite continues to ring and nothing happens.<br>> > As for video in working calls - the problem is with video from H324M<br>> > to SIP. Any ideas how to debug this?<br>> > <br>> > Can you provide a sample for using app_transcoder?<br>> > <br>> > Thanks,<br>> > Dan<br>> > <br>> > <br>> > <br>> > On Fri, May 9, 2008 at 1:27 PM, Sergio Garcia Murillo<br>> > <sergio.garcia@fontventa.com <mailto:sergio.garcia@fontventa.com>><br>> > wrote:<br>> > <br>> > Could you send me a file with the h245 and h223 logs? (enable<br>> > them by h324m debug level 4)<br>> > <br>> > The most probable cause is that you isdn provider is doing echo<br>> > cancelation on the line, it usually causes random problems like<br>> > this.<br>> > <br>> > The problem with video from SIP->H324M is that it has to be h263<br>> > QCIF at maximun 52 kbs, if your videophone is not able to set<br>> > this up, you'll need to use the app_transcoder module.<br>> > <br>> > Best regards<br>> > Sergio<br>> > <br>> > ----- Original Message -----<br>> > From: Dan Julius [mailto:dan.julius@gmail.com<br>> > <mailto:dan.julius@gmail.com>]<br>> > To: asterisk-video@lists.digium.com<br>> > <mailto:asterisk-video@lists.digium.com><br>> > Sent: Fri, 9 May 2008 12:25:17 +0300<br>> > Subject: Re: [Asterisk-video] call handshake fails<br>> > <br>> > Further info:<br>> > <br>> > - In the failed calls, the mobile phone never sends a<br>> > masterSlaveDetermination packet (according to the h223 logs)<br>> > - Asterisk sends the terminalCapabilitiesSet,<br>> > masterSlaveDetermination and<br>> > then continues to send OpenLogicalChannels.<br>> > <br>> > Is it OK to send OpenLogicalChannel before receiving a<br>> > masterSlaveDetermination?<br>> > <br>> > Thanks,<br>> > Dan<br>> > <br>> > On Fri, May 9, 2008 at 2:25 AM, Dan Julius <dan.julius@gmail.com<br>> > <mailto:dan.julius@gmail.com>> wrote:<br>> > <br>> > > Hi, Everybody,<br>> > ><br>> > > I'm new to this project, so I apologize if my questions<br>> > might have<br>> > > already been answered elsewhere.<br>> > > I am using a X-Lite, Asterisk 1.4.19, a Digium TE122 card,<br>> > and a Samsung<br>> > > Z720 phone.<br>> > ><br>> > > So far I have been able to make SIP-h234m calls (originating<br>> > at either<br>> > > side) with only partial success.<br>> > > - I only get video in one direction, from SIP to H324M. I've<br>> > read the posts<br>> > > stating that SIP->H324m is actually more problematic, so I'm<br>> > quite puzzled<br>> > > about this.<br>> > > - About 33% of the calls fail to negotiate a video<br>> > connection. After<br>> > > answering the call, nothing happens until I disconnect.<br>> > > The out-bound h223 log of a failed call is below. Does this<br>> > log indicate<br>> > > that Asterisk is sending terminalCapabilitySet multiple times<br>> > until it is<br>> > > acknowledged?<br>> > ><br>> > > 1 0.000000 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2<br>> > <http://2.2.2.2> H.245 terminalCapabilitySet<br>> > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet<br>> > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet<br>> > > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet<br>> > > masterSlaveDetermination masterSlaveDetermination<br>> > masterSlaveDetermination<br>> > > masterSlaveDetermination masterSlaveDetermination<br>> > masterSlaveDetermination<br>> > > masterSlaveDetermination masterSlaveDetermination<br>> > masterSlaveDetermination<br>> > > masterSlaveDetermination masterSlaveDetermination<br>> > masterSlaveDetermination<br>> > > masterSlaveDetermination masterSlaveDetermination<br>> > masterSlaveDetermination<br>> > > 2 0.000001 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2<br>> > <http://2.2.2.2> H.245 openLogicalChannel<br>> > > (generic) openLogicalChannel (generic) openLogicalChannel<br>> > (generic)<br>> > > openLogicalChannel (generic) openLogicalChannel (generic)<br>> > openLogicalChannel<br>> > > (generic) openLogicalChannel (generic) openLogicalChannel<br>> > (generic)<br>> > > openLogicalChannel (generic) openLogicalChannel (generic)<br>> > openLogicalChannel<br>> > > (generic) openLogicalChannel (generic) openLogicalChannel<br>> > (generic)<br>> > > openLogicalChannel (h263VideoCapability) openLogicalChannel<br>> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)<br>> > > openLogicalChannel (h263VideoCapability) openLogicalChannel<br>> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)<br>> > > openLogicalChannel (h263VideoCapability) openLogicalChannel<br>> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)<br>> > > openLogicalChannel (h263VideoCapability) openLogicalChannel<br>> > > (h263VideoCapability) openLogicalChannel (h263VideoCapability)<br>> > > openLogicalChannel (h263VideoCapability) openLogicalChannel<br>> > > (h263VideoCapability) multiplexEntrySend multiplexEntrySend<br>> > > multiplexEntrySend multiplexEntrySend multiplexEntrySend<br>> > multiplexEntrySend<br>> > > multiplexEntrySend multiplexEntrySend multiplexEntrySend<br>> > multiplexEntrySend<br>> > > multiplexEntrySend multiplexEntrySend<br>> > > 3 0.000002 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2<br>> > <http://2.2.2.2> H.245 multiplexEntrySend<br>> > > multiplexEntrySend multiplexEntrySend multiplexEntrySend<br>> > > terminalCapabilitySetAck terminalCapabilitySetAck<br>> > terminalCapabilitySetAck<br>> > > terminalCapabilitySetAck terminalCapabilitySetAck<br>> > terminalCapabilitySetAck<br>> > > terminalCapabilitySetAck terminalCapabilitySetAck<br>> > terminalCapabilitySetAck<br>> > > terminalCapabilitySetAck terminalCapabilitySetAck<br>> > terminalCapabilitySetAck<br>> > > terminalCapabilitySetAck terminalCapabilitySetAck<br>> > terminalCapabilitySetAck<br>> > > terminalCapabilitySetAck<br>> > > 4 0.000003 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2<br>> > <http://2.2.2.2> H223<br>> > > 5 0.000004 1.1.1.1 <http://1.1.1.1> -> 2.2.2.2<br>> > <http://2.2.2.2> H223<br>> > ><br>> > > Any pointers on how to debug this would be much appreciated.<br>> > ><br>> > > Thanks,<br>> > > Dan<br>> > ><br>> > > PS - This is really great work and I'm very impressed with<br>> > the project and<br>> > > hope that I will be able to contribute as well.<br>> > ><br>> > ><br>> > ><br>> > ><br>> > ><br>> > ><br>> > ><br>> > <br>> > <br>> > _______________________________________________<br>> > --Bandwidth and Colocation Provided by http://www.api-digital.com--<br>> > <br>> > asterisk-video mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-video<br>> > <br>> > <br>> > <br>> > <br>> > <br>> > ------------------------------------------------------------------------<br>> > <br>> > _______________________________________________<br>> > --Bandwidth and Colocation Provided by http://www.api-digital.com--<br>> > <br>> > asterisk-video mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-video<br>> <br>> _______________________________________________<br>> --Bandwidth and Colocation Provided by http://www.api-digital.com--<br>> <br>> asterisk-video mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-video<br><br /><hr />Connect to the next generation of MSN Messenger <a href='http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline' target='_new'>Get it now! </a></body>
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