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

2010-03-08 Thread 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








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Re: [asterisk-users] SIP to IAX to SIP

2009-10-19 Thread George Farris
On Mon, 2009-10-19 at 08:02 -0500,
asterisk-users-requ...@lists.digium.com wrote:
George Farris wrote:
>> I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs
>> very well.  On that machine I have a SIP phone.  I have configured a
>> netgear wgt634u with asterisk and a SIP phone and linked the two systems
>> together via IAX.  Audio from Ubuntu to netgear is not bad, audio from
>> netgear to ubuntu is unintelligible.  Any clues as to whether this will
>> work?  Configuration suggestions?  Is a 200MHz arm processor just too
>> small?
>what codec are you using?

I'm pretty sure it is GSM.

Cheers
George





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Re: [asterisk-users] SIP to IAX to SIP

2009-10-16 Thread Ivan Stepaniuk
George Farris wrote:
> I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs
> very well.  On that machine I have a SIP phone.  I have configured a
> netgear wgt634u with asterisk and a SIP phone and linked the two systems
> together via IAX.  Audio from Ubuntu to netgear is not bad, audio from
> netgear to ubuntu is unintelligible.  Any clues as to whether this will
> work?  Configuration suggestions?  Is a 200MHz arm processor just too
> small?
what codec are you using?

-- 
Iván Stepaniuk
Alba Fotónica S. L.
www.albafotonica.com


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[asterisk-users] SIP to IAX to SIP

2009-10-16 Thread George Farris

Hi all,

I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs
very well.  On that machine I have a SIP phone.  I have configured a
netgear wgt634u with asterisk and a SIP phone and linked the two systems
together via IAX.  Audio from Ubuntu to netgear is not bad, audio from
netgear to ubuntu is unintelligible.  Any clues as to whether this will
work?  Configuration suggestions?  Is a 200MHz arm processor just too
small?

Any help appreciated.


sip phone <--> wgt634u  <- iax -> ubuntu <--> sip phone
wireless   200MHz arm 3GHz AMDhardwired 100MB

 100MB lan between systems

Hope this is clear enough.


Cheers
George



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