[Freeswitch-users] g.722 --> SPEEX/16000 or SPEEX/32000 transcoding issue

[Bruce Hopkins]

Aha - I will try this now.  At the moment I only have SPEEX without anything
like @8000h or 16000h.

Thanks in advance, as I am sure you have spotted my noob error !

Bruce

On 9 February 2010 22:00, Brian West <brian at freeswitch.org> wrote:

> You need to allow SPEEX at 8000h,SPEEX at 16000h,SPEEX at 32000h
>
> /b
>
> On Feb 9, 2010, at 3:55 PM, Bruce Hopkins wrote:
>
> > Willdo,
> >
> > To clarify in brief though, the scenario which occurs and causes the call
> to fail is:
> >
> > SIP client  1 (g.722 enabled only ) -----> INVITE (with SDP offer: Media
> Attribute (a): rtpmap:9 g722/8000 ) -->FreeSWITCH
> >
> > ---> INVITE (with SDP offer including a bunch of codecs including rtpmap:
> rtpmap:98 SPEEX/8000 but crucially not including SPEEX/16000 or SPEEX/32000)
> >
> > ---> SIP client 2 (with only SPEEX/16000 or SPEEX/32000 enabled).
> >
> > The second SIP client does not get offered a codec it can accept, so SIP
> client 1 is sent a method 488 "Not Acceptable Here" message and the calling
> party gets directed to the voicemail for the other SIP client.
> >
> > By contrast, there is no problem calling SPEEX/32000 --> SPEEX/32000 or
> calling SPEEX/16000 --> SPEEX/16000.
> >
> > there is also no problem calling SPEEX/32000 --> g.722/8000.
> >
> > I am wondering if the problem is that FreeSWITCH is interpreting g.722 as
> being a narrowband (8kHz sample rate) codec, due to the historic anomaly of
> it presenting g722/8000 in the SDP even though it in fact uses 16kHz
> sampling, and for that reason not wanting to offer a 16kHz sample rate codec
> to the second SIP client?
> >
> > I suggest this as I also found trying to call alaw --> SPEEX/16000 does
> not work, for example.
>
>
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