[asterisk-ss7] Buffer full Audio lost.

[Anton]

Hi Asif!

BTW: Posting that is the list for any others who interested 
in audiolost problem

What I did now: I interconnected two asterisks with IAX2 - 
so the first one comes with SS7 and the second does VoIP - 
and IAX2 between them - Number of audiolosts semms much 
decreased (though some exists) - now I'm quite sure that 
asterisk REQUIRES internal jitter management to prevent TDM 
desyncronization. So desync always happens while using SIP 
as interconnection/OR any connection with any other SIP 
device. That's is a particular solution. Though - Now I see 
that with approx 20 simultaneous calls asterisk eats 15+ % 
of CPU - Seems jitterbuffer/and/or/PLC overhead. I'm sure 
that code could be tuned/replaced with better realization 
which will not require such a huge CPU power.

Anton.

On 12 April 2006 09:38, asif uddin wrote:
> Hi Anton,
>     Sorry for replying you so late. Actually i am out of
> station. I am travelling to some places now. Now
> regarding the chan_ss7 problem. As of my experience with
> audio lost problem, we have solved this problem by
> changing and modifying the application. We have not done
> any modification in chan_ss7. Hope  u r problem is
> solved. If not yet solved , i suggest u to see it from
> application and Dial Plan side.Because I am confident
> about ss7 part. And if U have solved it, please let me
> know how u have solved and what was the problem and why
> is ithappening with chan_ss7. Anton , Do u have T1.111
> Standard. Because i dont have it. Can u send it to me.It
> will be a great help  for me.
>
>
>
>
>   With Warm regards,
>   Asif.
>
> Anton <anton.vazir at gmail.com> wrote:
>   Hi Asif,
>
> Thanks for invitation for Yahoo, but I use Linux LICQ and
> not yahoo messenger, I'll try to set it up, but doubt
> that linux version supports voice :)
>
> Regarding the chan_ss7 - there is a data, and if look at
> the message it said "written 0 of 160" - so data exists.
> And audially such messages heared as clicks on the
> channels.
>
> I've tried to interconnect 2 asterisk boxes via SS7 and
> it does not shgw any losts - so it means that happen only
> when there is RTP transmission involved.
>
> Any ideas?
> Anton.
>
> On 9 April 2006 19:08, asif uddin wrote:
> > Hi Anton,
> > Sorry for the late reply. It was a weekend so iwent
> > out. I am not using chan_ss7 with any VoIP interface. I
> > am using it directly with the ss7 interface ( i got an
> > E1 from telco). See the problem what i feel is u r
> > trying to send something on say CIC=29 from chan_ss7. U
> > have allocated the channel successful;ly , but u r
> > unable to play the voice file (i.e there is nothing to
> > play on that channel) . Therefore in this case chan_ss7
> > (ss7_write) is trying to play the audio file , but
> > there is no audio file .and i think it is going into
> > infinite loop also.
> >
> > Asif
> >
> >
> >
> > Anton wrote:
> > Hi Asif!
> >
> > Did you use chan_ss7 terminated/originated calls with
> > VoIP? SIP/H323 729/723 codecs? I thing problem is
> > somewhere in desycronization...
> >
> > On 8 April 2006 21:17, asif uddin wrote:
> > > Hi Anton ,
> > > I have used chan_ss7 , both for incoming as well as
> > > outgoing w/o any problem.
> > >
> > > But once i got the similar kind of problem which u
> > > got and then I broke my head behind the ss7 code, but
> > > finally when we have modified the application (dial
> > > out) , the problem was solved and the application is
> > > running perfectly. can u tell me ? application u r
> > > using. so that we can proceed further .
> > >
> > > Asif
> > >
> > > Anton wrote: Hi Asif! Thanks for
> > > your mail.
> > >
> > > I use it as a gateway to cellular operator. I looked
> > > in the code - the message is returned by ss7_write
> > > function, which writes to a zaptel file descriptor.
> > > So it just unable to write there, since it's getting
> > > E_AGAIN from unix write(fd,...)
> > >
> > > than I looked at the zaptel code. That stream is
> > > received by zt_read() function. I did not dig further
> > > yet. May be you can.
> > >
> > > Now I'm going to do another test - 2 servers
> > > interconnected by 2xE1, and I'll pass 60 simultaneous
> > > calls with audio. And will see what will happen than.
> > >
> > > Regards,
> > > Anton
> > >
> > > On 8 April 2006 20:44, asif uddin wrote:
> > > > Hi Anton,
> > > > Thank u for sending the file, i think it is only
> > > > isup part (chan_ss7.c) , is there any progress in
> > > > mtp. Because i am good at mtp level , any way i
> > > > will see the isup part also. Now regarding the
> > > > Audio lost problem. I have faced the similar kind
> > > > of problem . Let me tell u that there is no problem
> > > > in the stack. Can u tell me ? application u r
> > > > using.
> > > >
> > > > >Did anyone resolved the issue of unability to
> > > > > write to zaptel FD? It's still an issue. My
> > > > > latest LIVE test showed that it happens with any
> > > > > codecs, when asterisk call another asterisk with
> > > > > g711 the only requirements - that there must be
> > > > > several on-going live calls. And as more calls is
> > > > > on - that happens more and more often. Audialy
> > > > > that heared as ticks in the sound. I do use
> > > > > Sangoma A102 card.
> > > > >
> > > > >Apr 8 19:39:38 NOTICE[21974]: chan_ss7.c:1884
> > > > > ss7_write: Write buffer full >on CIC=29 (wrote
> > > > > only 0 of 160), audio lost. Apr 8 19:39:38
> > > > > NOTICE[22014]: chan_ss7.c:1884 ss7_write: Write
> > > > > buffer full >on CIC=11 (wrote only 0 of 160),
> > > > > audio lost.
> > > >
> > > > Asif
> > > >
> > > >
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