[asterisk-ss7] chan_ss7 1.0.10 jitter buffer

[marek cervenka]

>>>> I do not understand the reason for having a jitter buffer in chan_ss7.
>>>> The audio is received in on a TDM line. Thus there is no jitter.
>>>>
>>> I think the same reason as chan_zap has a jitter buffer. As far as I
>>> know the other side of the conversation needs a jb.
>>
>> yes. if you terminate from SIP(outgoing call to PSTN) you need jb at
>> chan_ss7 side
>
> Ok. This is clear. But shouldn't the jitter buffer be implemented in
> chan_sip? How should chan_ss7 know if the audio is coming from a channel
> technology which causes jitter or not?

because of
Type: SIP (in channel)

(in one call are 2 channels bridged)

---------------------------------------
Marek Cervenka
=======================================