[asterisk-users] General network question regarding SIP and IAX2

[unserossi at aol.com]

Sounds great, thanks for your answer.
Do i need to set this on the trunk, the friend or on both?
 

 


 

 

-----Original Message-----
From: bruce bruce <bruceb444 at gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Sent: Fri, Jul 9, 2010 8:13 pm
Subject: Re: [asterisk-users] General network question regarding SIP and IAX2


The variable is canreinvite.
Please check on voipinfo. If canreinvite is enabled then only SIP signaling is passed through Asterisk and the media is not passed through Asterisk resulting in less bandwidth usage and probably less jitter buffer, etc,,,,if you are two phones are closer to each other than a round trip to Asterisk server.


On the flip side, you can't record these calls because no media is sent through Asterisk.


-Bruce


On Fri, Jul 9, 2010 at 1:48 PM,  <unserossi at aol.com> wrote:

Hi all,

i have a beginners question. How are SIP calls and IAX2 calls processed by Asterisk over the network?
What i mean is, is there a permanent connection required between the Asterisk Server and the clients or is the Asterisk Server only involved for lets call it the "routing"?