[asterisk-users] Audio data between concurrent SIP and PSTN

[Alex Balashov]

If I understand what you are asking correctly, no, all media streams - and
furthermore, their directional legs - are distinct.

IAX2 muxes media streams together into one logical connection.

SIP-driven RTP does not.

On Fri, August 29, 2008 5:46 pm, Allann Jones wrote:

> Hi. Are the audio streams returned by the user been shared between SIP and
> PSTN connections? I'm developing a speech recognition engine for Asterisk
> and I'm facing a problem where Asterisk is crashing when concurrent SIP
> and
> PSTN connections occur. I will read the code that implement that to
> understand it, but if anyone knows about that problem and can explain
> about
> it, thank you.
>
>
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-- 
Alex Balashov
Evariste Systems
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