asterisk users - Dec 2005 - Pls. explain what happens...

Hi everybody,
can anybody explain one thing: say we have 2 SIP phones(or H323) and one
Asterisk Box on one local network. The phone1 calls phone 2 via Asterisk and
phon3 answers: is the real conversation streaming thru the * box, or it's
going straigth from one phone to the other?

Regards and Happy New Year.

Mauro

Hello Mauro,

Tuesday, December 27, 2005, 9:26:54 AM, you wrote:

MZ> Hi everybody,
MZ> can anybody explain one thing: say we have 2 SIP phones(or H323) and one
MZ> Asterisk Box on one local network. The phone1 calls phone 2 via Asterisk
and
MZ> phon3 answers: is the real conversation streaming thru the * box, or
it's
MZ> going straigth from one phone to the other?

FOr what I know, It depends:

if you have reinvite set to "yes" in sip.conf, and there is no need of
format
transcoding, the audio stream goes directly from phone to phone.

To check that just unplug the ethernet cable from the asterisk server
while having a conversation between 2 phones: call should stay up.

Hope it helps!


-- 
Best regards,
 Alessio                            mailto:afoc@interconnessioni.it

On Tue, December 27, 2005 9:26, Mauro Zanin said:
> Hi everybody,
> can anybody explain one thing: say we have 2 SIP phones(or H323) and one
> Asterisk Box on one local network. The phone1 calls phone 2 via Asterisk
> and
> phon3 answers: is the real conversation streaming thru the * box, or
it's
> going straigth from one phone to the other?
>
> Regards and Happy New Year.
>
> Mauro
>
That depends on several factors, but basically:
CanReinvite = no => (*) always inbetween
CanReinvite = yes => If no NAT or other limiting factors (firewalls, etc.)
in place, phones will direct connect, otherwise (*) will handle the flow.

HTH

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