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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> --0-489812878-1171445129=:71204--
>
>
>
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