[sip-comm-dev] Got a Problem in sip communicator alpha version


#1

hi,
   
In Sip-Communicator alpha version we have got a problem accorfing to the following scenario.
  sip user A calls Sip user B through Asterisk with speex codec the callee could able to receive the voice but the caller could not received any voice.
  we have attached the log file with this mail.for quick reference we have given the sdp headers below.
  
v=0
o=root 7114 7114 IN IP4 192.168.2.33
s=session
c=IN IP4 192.168.2.33
t=0 0
m=audio 5000 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
   
  what is the "silenceSupp".. how to handle this problem..
   
  regards,
  Gokul

sip-communicator0(1).log (80.2 KB)

···

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#2

Hi,
Can you try using the latest build. as there were some issues with the
input buffer in alpha version and some times happens to not hear the
media . The media was received but not played by jmf and audio device.
And report does the problem remains.

damencho

···

On 2/14/07, M.Gokulakrishnan <smartgoki@yahoo.co.in> wrote:

hi,

In Sip-Communicator alpha version we have got a problem accorfing to the
following scenario.
sip user A calls Sip user B through Asterisk with speex codec the callee
could able to receive the voice but the caller could not received any voice.
we have attached the log file with this mail.for quick reference we have
given the sdp headers below.

v=0
o=root 7114 7114 IN IP4 192.168.2.33
s=session
c=IN IP4 192.168.2.33
t=0 0
m=audio 5000 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

what is the "silenceSupp".. how to handle this problem..

regards,
Gokul

________________________________
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Have a HUGE year through Yahoo! Small Business.
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#3

hi damian,
   
  thank u for ur quick reply..
   
  we will try out with your idea..
   
  first of all, whats the "silenceSupp" in SDP Headers..
   
  we are trying to implement the custom codecs like SPEEX and iLBC in SIP Communicator alpha.. To implement the custom codecs what we have to do..
  what are the necessary classes to be refer and the changes have to do in the sip communicator alpha version other than registering the custom codecs in the plugin manager.
   
  regards,
  Gokul
  
  Hi,
Can you try using the latest build. as there were some issues with the
input buffer in alpha version and some times happens to not hear the
media . The media was received but not played by jmf and audio device.
And report does the problem remains.

damencho

···

Damian Minkov <damencho@damencho.com> wrote:

On 2/14/07, M.Gokulakrishnan wrote:

hi,

In Sip-Communicator alpha version we have got a problem accorfing to the
following scenario.
sip user A calls Sip user B through Asterisk with speex codec the callee
could able to receive the voice but the caller could not received any voice.
we have attached the log file with this mail.for quick reference we have
given the sdp headers below.

v=0
o=root 7114 7114 IN IP4 192.168.2.33
s=session
c=IN IP4 192.168.2.33
t=0 0
m=audio 5000 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

what is the "silenceSupp".. how to handle this problem..

regards,
Gokul

________________________________
Get your own web address.
Have a HUGE year through Yahoo! Small Business.
--0-489812878-1171445129=:71204--

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#4

hi again,
silenceSupp is turned on - when there is now speech in the call no
media is send for the silence.
There such codecs in sip-communicator now. But still some issues exist
with speex codec. You can look at package
net.java.sip.communicator.impl.media.codec.

There some issues in jmf and sdp codes for codecs and number of
supported codecs in it.
But speex and ilbc have been already implemented.

damencho.

···

On 2/14/07, M.Gokulakrishnan <smartgoki@yahoo.co.in> wrote:

hi damian,

thank u for ur quick reply..

we will try out with your idea..

first of all, whats the "silenceSupp" in SDP Headers..

we are trying to implement the custom codecs like SPEEX and iLBC in SIP
Communicator alpha.. To implement the custom codecs what we have to do..
what are the necessary classes to be refer and the changes have to do in the
sip communicator alpha version other than registering the custom codecs in
the plugin manager.

regards,
Gokul

Damian Minkov <damencho@damencho.com> wrote:
Hi,
Can you try using the latest build. as there were some issues with the
input buffer in alpha version and some times happens to not hear the
media . The media was received but not played by jmf and audio device.
And report does the problem remains.

damencho

On 2/14/07, M.Gokulakrishnan wrote:
> hi,
>
> In Sip-Communicator alpha version we have got a problem accorfing to the
> following scenario.
> sip user A calls Sip user B through Asterisk with speex codec the callee
> could able to receive the voice but the caller could not received any
voice.
> we have attached the log file with this mail.for quick reference we have
> given the sdp headers below.
>
> v=0
> o=root 7114 7114 IN IP4 192.168.2.33
> s=session
> c=IN IP4 192.168.2.33
> t=0 0
> m=audio 5000 RTP/AVP 110 101
> a=rtpmap:110 speex/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> what is the "silenceSupp".. how to handle this problem..
>
> regards,
> Gokul
>
> ________________________________
> Get your own web address.
> Have a HUGE year through Yahoo! Small Business.
> --0-489812878-1171445129=:71204--
>
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> To unsubscribe, e-mail:
> dev-unsubscribe@sip-communicator.dev.java.net
> For additional commands, e-mail:
> dev-help@sip-communicator.dev.java.net
>

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