Asterisk is not a sip proxy but it *can* partly act as a sip proxy if
reinvites are enabled ( canreinvite=yes in sip.conf ) only then asterisk
connects 2 end points directly and does signalling between them .
Asterisk is a PBX now suppose u need to record all calls ..do conferencing
stuff  then rtp stream need to pass from asterisk (  openser cant do this
bcoz it just connects 2 endpoints  and only does signalling ) .. If you do
canreinvite=yes in sip.conf for both peers then asterisk does only
signalling ( also dial command should not have transfer parameters tT .. ) .
If both peers are behind NAT then asterisk reinvites may not work properly .

On 23/07/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:

Dear Edgar;

I am little bit confused, do u mean that asterisk does
not work in that way:

RTP (media) to be from the sournce to the destination
directly while signaling to be via asterisk?

So, what he parameter canreinvite is doing?

Regards,
------------
ITS
Ip Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460




Hello Asteriskers,

I'm confused about why Asterisk is not a SIP proxy and
why exactly
this can affect the performance of a large Asterisk
system.

I know that Asterisk acts as a useragent endpoint, but
my doubt is why
exactly Asterisk could overload the call flow if the
RTP voice stream
goes from the caller to the called party.

Does someone know how many calls or pencentaje that
could handle a SER
or OpenSER in comparison with Asterisk?







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