[Asterisk-Dev] codec renegotiation and fax [was: codec negotiation issue]

Daniel Bichara daniel at bichara.com.br
Thu Mar 24 17:29:01 MST 2005


Hi All,

So, if we make Asterisk start a renegotiation we could change the codec 
to G.711 during a call after we detected a fax. And we will not need to 
implement T.38 to make fax work between IAX or SIP devices. Cool.

Daniel

Michael Giagnocavo wrote:

>Well, if Asterisk already supports renegotiation... that's pretty cool then.
>I'd have to try it out with the PA168 phones, which are the only decent IAX
>devices on the market AFAIK (although some promising ones like the FarFon
>might be soon?).
>
>-Michael
>
>-----Original Message-----
>From: asterisk-dev-bounces at lists.digium.com
>[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Kevin P. Fleming
>Sent: Thursday, March 24, 2005 1:03 PM
>To: Asterisk Developers Mailing List
>Subject: Re: [Asterisk-Dev] codec negotiation issue
>
>Michael Giagnocavo wrote:
>
>  
>
>>Precisely because of that - support. It's not there today, which means
>>    
>>
>it's
>  
>
>>not in any devices out there today either. Also, it means it's not on any
>>providers today. So, we're talking a long time until it's universally
>>supported. However, the problem is today. And also applies to every other
>>channel, if their devices don't support renegotiation.
>>    
>>
>
>Pretty much every SIP and IAX device you can talk to today already 
>supports renegotiation during the call. The issue right now is that 
>Asterisk never initiates it, when it has information that would allow it 
>to do so.
>
>I could be wrong, but I suspect that when Asterisk is ready to start 
>doing this, it will 'just work' with all the providers and devices you 
>are referring to. The SIP and SDP protocols already fully support it, 
>and many devices already try to do it. (ever tried hitting "Conference" 
>on a Cisco 7940/7960 while in a G.729 call? it sends a re-INVITE trying 
>to switch the call to G.711)
>_______________________________________________
>Asterisk-Dev mailing list
>Asterisk-Dev at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-dev
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>_______________________________________________
>Asterisk-Dev mailing list
>Asterisk-Dev at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-dev
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>  
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20050324/5ee5d640/attachment.htm


More information about the asterisk-dev mailing list