SIP and IAX jitter buffer question

Asterisk Asterisk Support
1

Hi,

I have questions on the SIP jitter buffer implementation and IAX jitter buffer.

First, for SIP is it possible to force the jitter buffer for one peer only? I know it is on the receiving side only but I would like to force it on the receiving side on one peer only.

In addition to this, is it possible to generate silence rtp packets when there are too many frames received outside of the expected delay? The reason for this is that I don’t want the local phone to trigger its own jitter buffer if there are dropped frames which can’t be interpolated or buffered on the Asterisk jitter buffer.

My question for IAX jitter buffer is can we set the implementation to fixed similar to the SIP implementation or set a minimum jitter buffer? Can we also force to generate silence frame if there are too many missing frames which can’t be interpolated by the jitter buffer?

I’m using Asterisk 1.6.2

Thanks in advance!

J-F

2

Asterisk doesn’t normally use a jitter buffer for SIP. It might be possible to force it to do so, but this role is normally the responsibility of the phone.