[asterisk-dev] SIP <=> IAX - and RTP

Johansson Olle E oej at edvina.net
Mon Aug 25 04:40:06 CDT 2008


24 aug 2008 kl. 21.13 skrev Darren Sessions:

> I'm curious to know what the difference would be in how an RTP  
> stream in a SIP vs. IAX call is handled internally in Asterisk. I've  
> got an app_swift user that is complaining the app_swift module is  
> producing garbled audio on his IAX channels, and perfect audio on  
> the SIP channels. I was under the impression there wouldn't be a  
> difference on the RTP side.
>
> Any ideas before I tell him to troubleshoot his IAX setup?
>
FYI : IAX doesn't use RTP. Everything in a SIP to IAX call goes  
through the Asterisk core. Other than that, you have to deliver more  
information to get some help, like Asterisk version, and
please use the right mailing list - asterisk-users

Regards,
/O



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