[asterisk-users] Media Proxy Mode in Asterik: SIP and

[Alex Balashov]

Bilal,

On Fri, 13 Jul 2007, bilal ghayyad wrote:

> Please explain for me what do u mean exactly in "a la"
> in the following sentence u wrote it below?
>
> " in SIP, this can be done via
> "re-INVITEs" a la the canreinvite= option for SIP
> peers in sip.conf"

   It is an English colloquialism that enjoys wide currency;  stolen
from French, where it means "in the manner of" or "in the style of,"
for instance, "a la provençale" ("country style").  So, in this case
I mean that re-INVITEs are accomplished _by way of_ the 'canreinvite'
option.  Despite the nomenclature, 'canreinvite' in sip.conf actually
means not merely "CAN attempt to re-INVITE" but "WILL attempt to
re-INVITE as a point of default behaviour."

> Another thing, do u mean that it is easier (better) if
> we need H.323 endpoint to talk with SIP endpoint then
> we use full proxy (signaling + media) so that will
> solve miss compatibility issues?

   Well, an H.323 endpoint cannot speak directly to a SIP endpoint without 
a signaling gateway to convert the messages.  If you are referring purely
to passing RTP media, I suppose it does not matter, if you can broker that
handoff somehow between H.323 and SIP.  I am not sure if Asterisk has this
capability.

> By the way: Asterisk allow H.323 to talk with SIP?

   In principle, yes.  Asterisk can act as an H.323 gatekeeper, although
not as an endpoint (which is irrelevant to your purpose anyway).

   By far the easiest and cleanest solution is to run both the signaling
and the media through Asterisk, if you're making a SIP <-> H.323 call.
This is the approach least likely to cause any compatibility issues
since Asterisk is intermediating in the transaction, and I am not sure
if an alternative is even possible.

--
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : +1-678-954-0670
Direct : +1-678-954-0671