[Asterisk-Users] Why * try to codec translate when it can do without during codec negotiation.

[SamW]

I always like to use * to negotiate a codec which both sip clients
support. But * do not try to go in that direction, * try to be in the
middle and try to translate(code convert) the rtp stream, which will
deteriorate the call quality and it will also cost for license if we are
translating using G729 codec. 

Here is what happens on a test that I performed. I only list the fields
that I think are important. 

sip.conf

I have 2 sip clients sip-a and sip-b both cisco ATA-186 with sip-a set
to g711 (alaw/ulaw), and sip-b set to g729. If * request with a
particular codec then ATA can change to that codec irrespective of the
codec that is set as the preference. 


case 1
------
[sip-a]
allow=g729
disallow=all
allow=alaw

[sip-b]
disallow=all
allow=alaw
allow=g729

sip-a call sip-b Call get establish with 
sip-a(alaw) ---> * (codec-translate) --> sip-b(g729)

Note : order of preference under a context do not matter. Else 


case 2
------
[sip-a]
disallow=all
allow=g729

[sip-b]
disallow=all
allow=g729
allow=alaw

sip-a(g729) ---> * (codec-passthrough) --> sip-b(g729)
Note : Here sip-b-ATA is set to g729 and asterisk thinks that sip-a-ATA
cannot handle g711 and hence it request ATA to be g729 codec, and ATA
switch to g729 and work as expected. 


Issue with the asterisk in case 1 above is that it request from ATA 2
choices and sip-a-ATA chooses g711 and asterisk has to start translate
the coders, which is cost both quality, cpu resources and money.

Is there a solution how to set the coders so that it will avoid
transcoding when ever it can ? 

Is this a bug or is this how asterisk is implemented? But it is
essential for users to decide whether we give priority to transcoding
($$ to VoiceAge/Digium) or try to use passthrough when ever possible.