[asterisk-ss7] chan_ss7 1.0.10 jitter buffer

[marek cervenka]

>> Hi!
>>
>> 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

PSTN <---(chan_ss7 w/jb) Asterisk SS7 <----SIP---- SIP phone

in reverse direction is jb in the phone
PSTN --->(chan_ss7) Asterisk SS7 ----SIP----> (jb) SIP phone

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Marek Cervenka
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