[asterisk-users] G.722 problems with IAX

[Tim Panton]

On 4 Sep 2009, at 07:53, Armin Schindler wrote:

> On Thu, 3 Sep 2009, Tilghman Lesher wrote:
>> On Thursday 03 September 2009 02:47:05 Armin Schindler wrote:
>>> Hello,
>>>
>>> I try to move our asterisk installation (3 Asterisk servers in  
>>> different
>>> offices connected using IAX and a lot of SIP phones, as well as ISDN
>>> connections using CAPI) to use G.722 instead of G.711.
>>>
>>> Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one,  
>>> which solves
>>> the gain problem).
>>> So SIP-to-SIP and to ISDN there is no problem. G.722 itself works  
>>> and
>>> transconding to G.711 for ISDN also works good.
>>> But when I make a connection through IAX to another asterisk (having
>>> allow=g722 to really use G.722 in IAX) the voice is 'broken'.
>>>
>>> I also work on G.722 for twinklephone and encountered a special  
>>> thing about
>>> G.722: It has a sample rate of 16000, but it announced as 8000 in  
>>> SDP.
>>> Since I have similar problem with my G.722-twinkle implementation,  
>>> it looks
>>> like the RTP and/or jitterbuffer code has a problem with that.
>>> Did I miss something here or is this really a bug?
>>
>> You missed that the IETF has a typo in the specification, stating  
>> that G.722
>> is to be stated as 8000, even though it's 16000.  This will remain,  
>> due to
>> backwards compatibility concerns.  Please see RFC 3551, section  
>> 4.5.2.
>> http://www.apps.ietf.org/rfc/rfc3551.html#sec-4.5.2
>
> No, I didn't miss that. See my text.
> I mentioned this because I think this might be the reason of the  
> problem and
> the incorrect handling in jitterbuffer, if it is the jitterbuffer.  
> It is
> just a guess, since everything else seems to work good.
> The question is why does G.722 via IAX has problems.
> Is anyone using it and can say it works in his setup?
>
> Armin
>

I've got g722 running through 1.4.22.2 with the patch set that targets  
1.4.7

Calls from our java iax softphone come in as IAX2 in g722 and leave  
via SIP to a g722 conference service.
seems to work ok. No transcoding, recording etc, and the jitterbuffer  
is _off_ since it's a VoIP to VoIP call.

(a few folks used it on  the VUC conference this afternoon - anyone  
have problems ?).

Tim.

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk



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