[asterisk-users] SIP to IAX to SIP Jitterbuffer question

[Karl Fife]

Question:
If I am IAX trunking between 2 Asterisk instances, and ultimately connecting 
to SIP endpoints on BOTH ends of the call, can I let the ENDPOINTS do ALL 
the jitterbuffering, or must the iax-trunk do its own jitterbuffering?

I'm asking because I'm ignorant to the nuanced MECHANICS of the transport:

That is to say, if asterisk is passively sending voice frames from one 
protocol the other, then it clearly WOULD NOT matter if they go through the 
asterisk instance out-of-order.  The endpoint's local jitterbuffer can 
re-order the frames/packets.  Therefore in that scenario, it would seem that 
one could effectively eliminate the IAX jitterbuffer entirely and slightly 
decrease latency.

On the OTHER hand if the voice frames are being 'repackaged' by asterisk on 
new time bounaries, then naturally iax would need to do ALL of its own 
jitterbuffering to prevent incremental losses from out-of-order packets.

As I write this, it occurs to me that there may be a third option in which 
IT DOESN'T MATTER because there will be little or no out-of-order delivery 
within the local ethernet broadcast domain (to which each sip endpoint is 
connected), AND THEREFORE the IAX de-jittering would effectively cause the 
AUTOMATIC jitterbuffer on the endpoints to 'dry up' appropriately to near 
zero.

Could someon critique my logic or speak to this question?

Thanks!

-Karl