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

[Bruce Hopkins]

Hi,

Using the FusionPBX ISO (FreeSWITCH 1.0.4) I find I cannot make a call which
requires transcoding from g.722 (or other codec which declares 8kHz sampling
rate in the SDP) to SPEEX/16000 or SPEEX/32000.

If the calling extension uses only g.722, alaw, ulaw, etc, then only the
SPEEX/8000 narrowband variety of SPEEX is offered to the recipient extension
in the SDP of the SIP invite.

If the call is initiated the other way round - e.g. Client using SPEEX/32000
--> FreeSWITCH --> Client using g.722,  then the call is transcoded with no
problem.

I am wondering if this is the intended behaviour, to avoid transcoding
narrowband --> wideband.  However what I am finding is that transcodinig
g722/8000 (wideband) to SPEEX (wideband or ultrawideband) does not seem to
work.

I would be most grateful if anyone were able to let me know if there is a
configuration option I can set to alter this behavious and allow the full
range of SPEEX sampling rates to be offered in the SDP to the receiving
party, regardless of the codec used by the calling party.

Also, is this perhaps different in a more recent version?

Many thanks
Bruce
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