[Asterisk-Users] Avoiding transcoding

[Dr. Rich Murphey]

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com 
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> George Pajari
> Sent: Saturday, July 10, 2004 4:58 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Avoiding transcoding
> 
> Dr. Murphey:
> 
> > How can one specify that codec selection should avoid 
> transcoding if 
> > possible?
> 
> Asterisk only transcodes if the original audio stream needs 
> to be connected either to (a) another audio stream or (b) an 
> internal Asterisk function -- and the original audio stream 
> is encoded in a format that is not compatible with the other 
> audio stream or Asterisk function.
>
> > The reason I ask is that when enum lookup succeeds and the 
> destination
> only
> > accepts ULAW, the various transcodings seems to garble the audio if 
> > GSM, ILBC, etc. are allowed.
> 
> To illustrate the point above further, if you have a 
> destination that only accepts ULAW, and an audio stream in 
> GSM or ILBC, you must transcode. It is simply not an option 
> since ULAW is not GSM, Asterisk has to convert
> ("transcode") the ULAW samples into GSM samples and vice versa.

Is it possible to configure asterisk to
select a codec common to both channels?

Per your example, I tested a call from XTEN (sip) that
allowed ULAW or GSM, through asterisk, to a destination
that accepts only ULAW.

XTEN(SIP) -> asterisk -> SIP destination

It should be possible to avoid
transcoding, but asterisk shows this:

Peer             User/ANR    Call ID      Seq (Tx/Rx)   Format
216.234.116.184  1800246846  1b07a93b13f  00102/00000   ULAW  
172.16.0.100     xten        EEE6B555-5B  00101/09156   GSM  

Because the destination is found by enum lookup, it's not
possible to know which protocol or codec in advance.

Thank very much for your explanations.

Cheers,
Rich