<div dir="ltr">Hi, Sergio,<br><br>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.<br>
<br>Turns out that increasing the retransmit delay from 20 to 2000 in 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><div class="gmail_quote">
On Thu, May 15, 2008 at 4:42 PM, Dan Julius <span dir="ltr"><<a href="mailto:dan.julius@gmail.com" target="_blank">dan.julius@gmail.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;">
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 discussion public?<br><br> Thanks for looking into this.<br><font color="#888888"><br>
Dan<br>
<br></font><div class="gmail_quote"><div>---------- Forwarded message ----------<br>From: <b class="gmail_sendername">Dan Julius</b> <<a href="mailto:dan.julius@gmail.com" target="_blank">dan.julius@gmail.com</a>><br>
Date: Fri, May 9, 2008 at 2:07 PM<br>
Subject: Re: [Asterisk-video] call handshake fails<br></div><div><div></div><div>To: Development discussion of video media support in Asterisk <<a href="mailto:asterisk-video@lists.digium.com" target="_blank">asterisk-video@lists.digium.com</a>><br>
<br><br>Hi,<br>
<br>Attached are logs for a call that failed. After answering the call 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 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><div class="gmail_quote"><div><div></div><div>On Fri, May 9, 2008 at 1:27 PM, Sergio Garcia Murillo <<a href="mailto:sergio.garcia@fontventa.com" target="_blank">sergio.garcia@fontventa.com</a>> wrote:<br>
</div></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div>Could you send me a file with the h245 and h223 logs? (enable them by h324m debug level 4)<br>
<br>
The most probable cause is that you isdn provider is doing echo cancelation on the line, it usually causes random problems like this.<br>
<br>
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.<br>
<br>
Best regards<br>
<font color="#888888">Sergio<br>
</font><div><div></div><div><br>
----- Original Message -----<br>
From: Dan Julius [mailto:<a href="mailto:dan.julius@gmail.com" target="_blank">dan.julius@gmail.com</a>]<br>
To: <a href="mailto:asterisk-video@lists.digium.com" target="_blank">asterisk-video@lists.digium.com</a><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, 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 <<a href="mailto:dan.julius@gmail.com" target="_blank">dan.julius@gmail.com</a>> wrote:<br>
<br>
> Hi, Everybody,<br>
><br>
> I'm new to this project, so I apologize if my questions might have<br>
> already been answered elsewhere.<br>
> I am using a X-Lite, Asterisk 1.4.19, a Digium TE122 card, and a Samsung<br>
> Z720 phone.<br>
><br>
> So far I have been able to make SIP-h234m calls (originating at either<br>
> side) with only partial success.<br>
> - I only get video in one direction, from SIP to H324M. I've read the posts<br>
> stating that SIP->H324m is actually more problematic, so I'm quite puzzled<br>
> about this.<br>
> - About 33% of the calls fail to negotiate a video 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 log indicate<br>
> that Asterisk is sending terminalCapabilitySet multiple times until it is<br>
> acknowledged?<br>
><br>
> 1 0.000000 <a href="http://1.1.1.1" target="_blank">1.1.1.1</a> -> <a href="http://2.2.2.2" target="_blank">2.2.2.2</a> H.245 terminalCapabilitySet<br>
> terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet<br>
> terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet<br>
> terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet<br>
> masterSlaveDetermination masterSlaveDetermination masterSlaveDetermination<br>
> masterSlaveDetermination masterSlaveDetermination masterSlaveDetermination<br>
> masterSlaveDetermination masterSlaveDetermination masterSlaveDetermination<br>
> masterSlaveDetermination masterSlaveDetermination masterSlaveDetermination<br>
> masterSlaveDetermination masterSlaveDetermination masterSlaveDetermination<br>
> 2 0.000001 <a href="http://1.1.1.1" target="_blank">1.1.1.1</a> -> <a href="http://2.2.2.2" target="_blank">2.2.2.2</a> H.245 openLogicalChannel<br>
> (generic) openLogicalChannel (generic) openLogicalChannel (generic)<br>
> openLogicalChannel (generic) openLogicalChannel (generic) openLogicalChannel<br>
> (generic) openLogicalChannel (generic) openLogicalChannel (generic)<br>
> openLogicalChannel (generic) openLogicalChannel (generic) openLogicalChannel<br>
> (generic) openLogicalChannel (generic) openLogicalChannel (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 multiplexEntrySend<br>
> multiplexEntrySend multiplexEntrySend multiplexEntrySend multiplexEntrySend<br>
> multiplexEntrySend multiplexEntrySend<br>
> 3 0.000002 <a href="http://1.1.1.1" target="_blank">1.1.1.1</a> -> <a href="http://2.2.2.2" target="_blank">2.2.2.2</a> H.245 multiplexEntrySend<br>
> multiplexEntrySend multiplexEntrySend multiplexEntrySend<br>
> terminalCapabilitySetAck terminalCapabilitySetAck terminalCapabilitySetAck<br>
> terminalCapabilitySetAck terminalCapabilitySetAck terminalCapabilitySetAck<br>
> terminalCapabilitySetAck terminalCapabilitySetAck terminalCapabilitySetAck<br>
> terminalCapabilitySetAck terminalCapabilitySetAck terminalCapabilitySetAck<br>
> terminalCapabilitySetAck terminalCapabilitySetAck terminalCapabilitySetAck<br>
> terminalCapabilitySetAck<br>
> 4 0.000003 <a href="http://1.1.1.1" target="_blank">1.1.1.1</a> -> <a href="http://2.2.2.2" target="_blank">2.2.2.2</a> H223<br>
> 5 0.000004 <a href="http://1.1.1.1" target="_blank">1.1.1.1</a> -> <a href="http://2.2.2.2" target="_blank">2.2.2.2</a> 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 the project and<br>
> hope that I will be able to contribute as well.<br>
><br>
><br>
><br>
><br>
><br>
><br>
><br>
<br>
</div></div><br></div></div>_______________________________________________<br>
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">http://www.api-digital.com--</a><br>
<br>
asterisk-video mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-video" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br></blockquote></div><br>
</div></div></div><br>
</blockquote></div><br></div>