[asterisk-users] pjsip: asterisk can't decide which codec to use
jcolp at digium.com
Fri May 12 13:49:40 CDT 2017
On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote:
> If I'm doing exactly the same call originated with another extension,
> there can't be seen these frequent changes. But the strange thing is,
> that in both cases the part between extension and asterisk doesn't show
> any codec changes ... .
> Deeper investigations show, that if the conference (callee) sends the
> first rtp package (-> g711 - should be g722), things are going choppy,
> if the extension (caller) sends the first package (g722), things are
> running stable.
> Any idea to convince asterisk always to use the first codec of ok sdp
> or how to convince asterisk to put only one codec to ok sdp (the first).
This is not currently an option in chan_pjsip but I'd suggest filing an
issue for this scenario with all available information.
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