Steve Totaro wrote:
On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin <[EMAIL PROTECTED]> wrote:
We run asterisk to handle incoming DIDs and we have observed
inefficient Codec Translation.
Here is the scenario
[DID Vendor] ---------------------------> [Asterisk ]
------------------------> External GW [G729]
|
|-------------------> External GW [iLBC]
Our DID vendor and asterisk box supports both ilbc & g729. However,
our external gateway termination supports either ilbc or g729 (and not
both) and depending on users location, we terminate it on either
gateway.
Since DID and asterisk box supports both the codecs, we assumed that
asterisk will appropriately select codecs depending on where we
terminate the call so that no codec translation happens. However, this
seems to be an incorrect assumption and we see that different codecs
get selected on two legs which leads to quality drop and extra CPU
cycles.
May be we are doing something wrong. Pls suggest what we are doing
wrong. Below is asterisk configuration.
[did]
type=friend
host=xxx
canreinvite=yes
disallow=all
allow=g729
allow=ilbc
[gw1]
type=friend
host=xxx
canreinvite=yes
disallow=all
allow=g729
[gw2]
type=friend
host=xxx
canreinvite=yes
disallow=all
allow=ilbc
Thanks
Jim
Why don't you allow=g729 only on all entries. Maybe I have misread
your email but I interpret what you wrote to mean that all endpoints
support g729
I may be wrong but I understood the situation as the DID supplier
supports either g.729 or ilibc, but the user has 2 locations that calls
are routed to. One location supports iLibc only, the other supports
g.729 only. What they seem to be trying to accomplish is to get the DID
<-> Asterisk leg to use the same codec as the Asterisk <-> Remote
Location leg. I think the problem is going to be that the call has to
be established to the Asterisk box before a destination can be
selected. The DID and Asterisk Box are going to negotiate the first
available common codec before doing anything else, including setting a
destination. Since you can't change a codec once a call has been
established you're always going to end up with calls to one of the 2
remote locations being transcoded.
The only solution I could think of would be if there was some way to
identify which incoming calls were going to be routed to which location
and set the codec accordingly. To do that, you'd either have to have 2
different DID's or some other massively more complicated mechanism.
Forcing a reinvite (Is that even possible?) would be the only other
long-shot I could think of.
Good luck,
Brent
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