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

[Bruce Hopkins]

Hi again Brian,

I can confirm that having followed your advice, it now works perfectly.

Many, many thanks for your extraordinarily quick and comprehensive help with
this.

Best wishes
Bruce

On 9 February 2010 22:13, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> 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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>
>
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