[asterisk-dev] chan_datacard and RTCP

santosh chintalwar santoshchintalwar at gmail.com
Sun Jan 17 13:49:27 CST 2010


There might be ACK issue.
If ACK is not reached to the Phone then it will terminate the call after
retransmission timeout.
Some time ago I got this issue with SJPhone but not with X-Lite.

Santosh Chintalwar
+91 9949695124


On Mon, Jan 18, 2010 at 1:02 AM, Steve Totaro
<stotaro at asteriskhelpdesk.com>wrote:

>
>
> On Sun, Jan 17, 2010 at 1:05 PM, Olle E. Johansson <oej at edvina.net> wrote:
>
>>
>> 17 jan 2010 kl. 17.29 skrev Artem Makhutov:
>>
>> > Hello,
>> >
>> > I am writing chan_datacard to be able to use Huawei UMTS datacards as a
>> GSM gateway with Asterisk.
>> >
>> > Some users have complained that their calls are disconnected after
>> exactly 30 seconds.
>> >
>> > After some investigation it came out that they were using X-Lite as
>> their sofphone and that X-Lite is disconnecting calls after 30 seconds if it
>> does not receive RTCP packages in that time.
>> >
>> > So the question is how can I get asterisk to send RTCP packets to
>> X-Lite?
>> >
>> > Must I add something to chan_datacard to generate RTCP packets or should
>> asterisk generate RTCP packets by its own?
>> > I have no clue about RTCP. Can somebody give me some hints at what I
>> should look at?
>>
>> Asterisk generates RTCP unless there's an p2p RTP bridge, which I guess
>> won't happen if you place a call from a SIP channel to your own channel.
>> Turn on "rtcp debug" in the CLI and you'll get detailed information about
>> RTCP messaging.
>>
>> I have never heard anything about X-lite disconnecting because there's no
>> RTCP reports, that's very odd.
>>
>> For a bit more information about RTCP, I suggest that you read my blog
>> about an on-going project to enhance RTCP support in Asterisk:
>>
>> http://www.voip-forum.com/asterisk/2010-01/measuring-voice-quality-asterisk/
>>
>> /O
>> --
>
>
> Not sure if it is related, but 30 seconds disconnect rings a bell if you
> don't Answer() in some circumstances.
>
> Thanks,
> Steve T
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20100118/29a975af/attachment.htm 


More information about the asterisk-dev mailing list