[asterisk-users] G.722 problems with IAX

[Armin Schindler]

On Fri, 4 Sep 2009, Tim Panton wrote:
> 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.

Yes, when is disable jitterbuffer (I had even forcejitterbuffer=yes in 
iax.conf and jbforce=yes in sip.conf), it works here too.

I don't know how the jitterbuffer is doing it, but could it be possible that
the RTP info of g722 (stated 8000 but it actually 16000) is confusing the 
jitterbuffer?

Armin