[asterisk-dev] Jitter Buffer (patch included)

[John Lange]

On Wed, 2006-10-18 at 22:26 +0200, Martin Vít wrote:
> John Lange wrote:
> > I'm trying to test the new jitter buffer in 1.4beta2 but have not been
> > able tell if its working.
> >
> > In the general section of sip.conf I have the following:
> >
> > jbenable = yes
> > jbforce = yes
> > jbmaxsize = 200
> > jbresyncthreshold = 1000
> > jbimpl = adaptive
> > jblog = yes 
> >
> > Here is the topology:
> >
> > T1 PSTN <--> Asterisk 1.2 <-> Asterisk 1.4b2 <-> Sip Phone
> >
> > I'm trying to get the jb working between Asterisk 1.4b2 and the Sip
> > Phone (which is a Linksys SPA942).
> >   
> Sip phone has its own jitterbuffer. Dejjiterring is needed for T1 
> termination.

The SIP phone has a jb for the packets it receives. If you are making a
call between two jb enabled endpoints then you don't need an Asterisk
JB.

In any other scenario you would need an Asterisk SIP JB which is what
I'm trying to do.

> Also jbenable is not correct it is jb-enabled.

No, it is jbenable etc. as I have quoted above. In respect to that I
have created this very small patch which displays the settings when sip
is reloaded.


--- asterisk-1.4.0-beta2/main/abstract_jb.c     2006-08-29 15:50:36.000000000 -0500
+++ /usr/src/asterisk-1.4.0-beta2/main/abstract_jb.c    2006-10-18 16:55:23.000000000 -0500
@@ -594,6 +594,9 @@
                return -1;
        }

+    if (option_verbose > 2)
+        ast_verbose(VERBOSE_PREFIX_3 "%s = '%s'\n", name, value);
+
        return 0;
 }


So using the above I have confirmed that the settings are being
recognized but still I have no logging or any indication on the console.

Is there some other requirement to get it working?

I tested the JB earlier from SVN before 1.4b2 was released and using the
same topology (PSTN <-> ASTERISK <-> ASTERISK <-> SIP PHONE) I was able
to see some JB information so I'm wondering if the JB is now
malfunctioning somehow?

John