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.
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