[Asterisk-Dev] SIP RTP JitterBuffer in Asterisk

[Zoa]

Can you try it with a playback of some file ? the milliwatt only gives
audio packets when it receives audio, which might give strange results,
we will reproduce your tests in the lab tomorrow.

Zoa

Matt Hess wrote:

> # gcc -v
> Reading specs from /usr/lib/gcc-lib/i386-unknown-openbsd3.6/2.95.3/specs
> gcc version 2.95.3 20010125 (prerelease, propolice)
>
> my results:
> An unpatched head system calling an external milliwatt number is nice
> and smooth. (good benchmark call)
>
> A head system with the latest patch on the same call is horrible. I've
> tried using both the default variables/settings and modifying them to
> no avail. Turning the jitter buffer off in sip.conf yields the same
> smooth call.
>
> in tinkering further with the settings in sip.conf:
>
> usejb=yes
> jbsize=300
> forcejb=no
> jblog=yes
>
> yields a good call but I note that I get the following log message:
> -- ***[RTP JB LOG]*** Detected bridged channel that can accept jitter.
> Disabling the jitterbuffer.
>
> with:
> usejb=yes
> jbsize=300
> forcejb=yes
> jblog=yes
>
> message log:
> -- ***[RTP JB LOG]*** Jitterbuffer created and started. jb_alloc=1.
> -- ***[RTP JB LOG]*** Jitterbuffer created and started. jb_alloc=2.
>
> and again the call is complete crap..
>
>
> The call itself is a gsm codec call transcoded to ulaw.. gsm on the
> external side and ulaw going to a bt-100 sip phone. The asterisk
> server has 2 network interfaces.. one public and one private. The
> bt-100 is on private ip space.
>
>
> Slav Klenov wrote:
>
>> What version of gcc you are using? With 3.4.4 this compile pretty
>> well. Actualy its a C++ style and some more strict compilers
>> (including some gcc versions) generate an error.
>>
>>
>> Matt Hess wrote:
>>
>>> I get a compile error.. did a fresh cvs get of asterisk and applied
>>> patch cleanly.
>>>
>>> gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
>>> -Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT
>>> -D_GNU_SOURCE  -O6 -march=i386  -pthread
>>> -fomit-frame-pointer    -c -o rtp.o rtp.c
>>> rtp.c: In function `rtp_jb_destroy':
>>> rtp.c:877: syntax error before `int'
>>> rtp.c:882: `len' undeclared (first use in this function)
>>> rtp.c:882: (Each undeclared identifier is reported only once
>>> rtp.c:882: for each function it appears in.)
>>> gmake: *** [rtp.o] Error 1
>>>
>>>
>>> Slav Klenov wrote:
>>>
>>>> New SIP jitterbuffer patch against today cvs-head is available on
>>>> mantis:
>>>>
>>>> http://bugs.digium.com/view.php?id=3854
>>>>
>>>>
>>>> Slav
>>>>
>>>> Olle E. Johansson wrote:
>>>>
>>>>> Matt wrote:
>>>>>
>>>>>
>>>>>> Hi,
>>>>>> I heard talk that there was a SIP RTP JitterBuffer which was
>>>>>> either in
>>>>>> asterisk CVS, or was being made as a patch here on the dev
>>>>>> list.   Can
>>>>>> anyone confirm this or deny it?  And if it exists, what is the
>>>>>> current
>>>>>> status of it?
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> At this point, a lot of work is going on to fix this. We do not
>>>>> know if
>>>>> we can fix it in due time for 1.2. There are non-working patches
>>>>> in the
>>>>> bug tracker, but rumour (a less than one hour old rumour) tells me
>>>>> that
>>>>> something is working out there. Nothing is included in CVS at this
>>>>> point.
>>>>>
>>>>> A bit longer answer than "no" this second time :-)
>>>>>
>>>>> /O
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>>>>>
>>>>
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