[asterisk-ss7] chan_ss7 + Astersik JitterBuffer

Patrick asterisk-list at puzzled.xs4all.nl
Sun Jun 25 01:50:52 MST 2006

On Sun, 2006-06-25 at 06:30 +0530, Mr.Surender Reddy wrote:
> Hello sir,
>             I have audio lost problem in ss7 is there any way that i
> can over come this.I have tried to what ever i could do which is said
> in all the forums like increase the jitter buffer 
> 4*160 like this and Clocking and all but iam unable to get a way that
> i can over come this problem.If any one could give me some help that
> would be a great help for me sir.When there is no audio lost the
> Quality is just like a PSTN and there is no comparision PSTN and VOip
> call when the audio lost comes the voice gets choppy and gitchhy.If
> anyone can give me a solution on commercial basis also is most
> welcome.
> regards
> surender

Hello Surender,

First you need to compile chan_ss7 with -DAST_JB added to the CFLAGS of
the chan_ss7 Makefile if you have built Asterisk 1.2.x with the
jitterbuffer patch applied. 

I'm not sure if chan_ss7 supports the jitterbuffer. I was told that the
jitterbuffer patch is specifically for SIP-SIP calls and ZAP-SIP/SIP-ZAP
calls and I don't know if the normal ZAP channels are used for SS7 too.

Perhaps you could ask the folks at Sifira if they can implement the
jitterbuffer on SS7 channels if you pay for it. Or ask the original
author slav [a.t] securax.org.

Alternatively try building Asterisk without the jitterbuffer patch and
see what happens.


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