[Asterisk-Users] rtp channels not through asterisk

[Alexander Lopez]

Look at canreinvite= in the sip.conf.

 

If you 'remove' Asterisk from the stream them you are using Asterisk
more like a Proxy and less like a PBX. If this is the case and you want
to support 'tons' of users look at something like SER.  Asterisk is not
a Sip proxy but rather a PBX and Media transcodeing gateway 

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of bijan
Sent: Thursday, December 23, 2004 5:46 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] rtp channels not through asterisk

 

In wiki pages it is stated that The audio channels (RTP) may go directly
from phone to phone or may go through Asterisk's media bridge.

Currently with my settings, I notice that all rtp's are passing through
my asterisk. How could I achieve that they go directly from phone to
phone?  I assume this way, my machine will have less load and therefore
could handle more calls.

 

regards

Bijan Karimi

 

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