[Asterisk-Dev] Per-call codec preference

[Paul]

Michael,

I have only been running * for 10 days now so please excuse any 
ignorance or stupidity in my comments.

1) It would definitely be very helpful for testing because I currently 
create extensions for testing providers. A typical test of outbound 
providers is that I use kphone dial 9d0p where d is the destination and 
p is the provider. I dial 9203 to call my office pots line using 
voicepulse. I usually put the line on hold to play some Hendrix or 
Joplin and then transfer it to the echo test. If you add this patch, I 
could change that to 9dcp where c is the codec to use. I have all these 
test extensions in an include file. Next thing I should do is write a 
macro to generate them.

2) I suppose it could also be used to implement a mechanism for 
dynamically adjusting to call volumes. I can easily move my * test setup 
between cable modem, adsl, sdsl and T-1 lines that I have control over. 
The average soho user does not have that option. I have already seen 
that many of those soho setups might want to change codecs as soon as 
they open a second pipe. My experiments with a callback system(so I can 
get unlimited calling from my home in the backwoods or from my cell 
phone with free incoming) indicate that 2 ulaw channels only work well 
on the T-1's from backbone providers. My normal usage when at the office 
is one call at a time to a softphone. That works fine even while 
downloading tarballs and using rsync to update mirrors. When I use the 
second channel the loss in quality is obvious. So I definitely would 
want to try changing codecs when doing the callback and also for any 
calls I place during the callback session.

3) Allowing users to dial *xx for per-call codec changes would be a nice 
feature.

4) Some of my tests so far indicate that using a high-quality codec to 
call my cell number is a waste of bandwidth. I have set it up as an 
extension and use a voip provider to call it. I use the same account for 
most of my outgoing calls so per-call codec negotiation would be nice here.

- Paul

Michael Giagnocavo wrote:

>Hi there,
>
>	I'm thinking of adding patches to allow per-call codec negotiation,
>so that the codec can be set via the dialplan ("at runtime"), instead of
>just when defining peers ("design time").
>
>	My idea would be to add a channel variable that could be a codec
>("g729", "ulaw", etc.), "current", to mean whatever codec is currently in
>use, or null/"default", for whatever would normally work.
>
>	Any comments or suggestions on this?
>
>-Michael
>
>
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