Scenario:
IP Phone --> SIP Server --> Asterisk --> (Voice Mailbox)
(G723.1, G729) -> (GSM) -> (GSM, G729) GSM coded file
Is ipphone with codec g723.1 and g729 cabable to retrived voice mail from the above picture? If not, please explain a little more.
Thanks a lot.
Innocent Evil wrote:
IP Phone --> SIP Server --> Asterisk --> (Voice Mailbox)
(G723.1, G729) -> (GSM) -> (GSM, G729) GSM coded file
The SIP server (if it's a proxy like SER) doesn't matter for codec negotiation, because it just does the SIP signalling.
So if your phone supports G.723.1 and G.729, then Asterisk has to support one of these (for G.729 you need a license from Digium, G.723.1 is only supported in pass-thru-mode AFAIK, so not usabe here).
Andy
Thanks for answering, Andy.
So the key point is, both end need to have same codec supported. It doesn't matter how many system/server/broker are between those two end, right?
Thanks again..
-----Original Message----- From: andreas.granig@inode.info Sent: Mon, 01 Aug 2005 20:01:19 +0200 To: innocent.evil@inbox.com Subject: Re: [Serusers] codec question
Innocent Evil wrote:
IP Phone --> SIP Server --> Asterisk --> (Voice Mailbox)
(G723.1, G729) -> (GSM) -> (GSM, G729) GSM coded file
The SIP server (if it's a proxy like SER) doesn't matter for codec negotiation, because it just does the SIP signalling.
So if your phone supports G.723.1 and G.729, then Asterisk has to support one of these (for G.729 you need a license from Digium, G.723.1 is only supported in pass-thru-mode AFAIK, so not usabe here).
Andy
Innocent Evil wrote:
So the key point is, both end need to have same codec supported. It doesn't matter how many system/server/broker are between those two end, right?
Well, all systems between, which decode the RTP stream on one call leg and encode it on the other one (like session border controllers, PSTN gateways etc.), have to support the codec of their peer on the corresponding call leg.
If they are pure SIP proxies, they don't get involved into the codec negotiation.
Andy