[pjsip] Distorted audio with g722

Saul Ibarra Corretge saul at ag-projects.com
Wed Dec 9 03:36:22 EST 2009

Hi Nanang,

On Dec 9, 2009, at 3:13 AM, Nanang Izzuddin wrote:

>> We got some feedback from Kevin Fleming (Asterisk) and Steve Underwood (author of SpanDSP). Both Asterisk and FreeSWITCH use Steve's g722 implementation and according to tho their answers, that g722 implementation doesn't shift those last 2 bits, instead it uses the first 14.
> Were you talking about this thread post [1]? Well, to me, it sounds
> that r2342 has just got another confirmation :) It is exactly what
> r2342 does.

Right, that's what I thought at first, but after reading [2] what I understood is that Asterisk just cares about the first 14bits, but pjsip shifts 2bits so the first 2 are not used and in Asterisk the upper 2 are the ones not used. It's the first time I'm dealing with this low-level codec issues, so maybe I got everything wrong :)

>>  - Reverted r2342 and played back a g722 file in Asterisk 1.6.2. Audio was ok.
> Hmm, dunno. Seems that Asterisk experienced same issue and the
> Asterisk version you were using seems to include the fix, as discussed
> in this thread [2]. Did the g722 file generated by Asterisk with g722
> fix included?

I used latest checkout of the 1.6.2 branch which seems to include some related fix.

> As so far, it seems that the problem is not in r2342 (please also note
> that r2342 also works great with VoiceAge implementation), wouldn't it
> be wiser to report/confirm this issue to ZipDX first?

You're right, we'll report it to ZipDX also and I'll also try to get some feedback from someone using a g722 capable hardphone to see what seems like the most interoperable option.

Best regards,

Saul Ibarra Corretge
AG Projects

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