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 <[email protected]> To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> 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, <[email protected]> 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"? >From my understanding SIP s used to "find" the "way" to the remote party and >the voice is transferred over RTP directly from client to client without >permanently involving the Server. IAX seems to do all in one, the "routing" and the transport of the voice. Is that correct? Why i am asking this? Lets say i have one Asterisk running in London and another one in Paris. Both are connected via IAX2 trunk over a WAN connection. User A is registered on the server in London. User B is registered on the server in Paris. Now User A is visiting User B in Paris and both have call with each other. Is the voice data routed from user A to Asterisk in London and then back via IAX2 to the server in Paris and the to user B? Or is there a direct connection between them and no WAN traffic is produced? And is there a difference between using either SIP or IAX as client protocol in that case? I hope i explained well what i meant. Thanks in advance for answers. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
