[jitsi-dev] Audio after reinvite sip


#1

Hi!
Do I have re-send to dev@jitsi.org questions that have been sent to
dev@jitsi.java.net?

···

2013/5/15 Emil Ivov <emcho@jitsi.org>

Hey Oleg,

Can you show us the wireshark dumps?

Emil

On 15.05.13, 00:41, Oleg Odinec wrote:
> Hello!
> I use Jitsi as sip phone.
> I receive a Call with first audio codec g711A, and I send response OK
> with first audio codec G711A,
> but when I enable video - it send REINVITE with first codec G722 and I
> get OK from SIP Server with first codec G722, whereupon I can not hear
> received audio.
> Wireshark shows: that after REINVITE rtpstreams are transmitting in
g722.
> My interlocutor hears me perfectly , but I do not hear him.
> Follows from the above that Jitsi handles sending RTP in changed codec,
> but it does not work with received audio.
>
> As well: After change the codec, SIP Server does not change SSRC and
> sends stream in new codec with old SSRC. maybe it is cause of the issue?
>
> Thanks!
>
> --------
> Best regards,
> Oleg

--
https://jitsi.org


#2

Hey Oleg,

Just had a look at the logs. It seems that after the re-INVITE the
server isn't sending any more media in your direction.

Hope this helps,
Emil

···

On 16.05.13, 09:20, Oleg Odinec wrote:

Hi!
Do I have re-send to dev@jitsi.org <mailto:dev@jitsi.org> questions
that have been sent to dev@jitsi.java.net <mailto:dev@jitsi.java.net>?

2013/5/15 Emil Ivov <emcho@jitsi.org <mailto:emcho@jitsi.org>>

    Hey Oleg,

    Can you show us the wireshark dumps?

    Emil

    On 15.05.13, 00:41, Oleg Odinec wrote:
    > Hello!
    > I use Jitsi as sip phone.
    > I receive a Call with first audio codec g711A, and I send response OK
    > with first audio codec G711A,
    > but when I enable video - it send REINVITE with first codec G722 and I
    > get OK from SIP Server with first codec G722, whereupon I can not
    hear
    > received audio.
    > Wireshark shows: that after REINVITE rtpstreams are transmitting
    in g722.
    > My interlocutor hears me perfectly , but I do not hear him.
    > Follows from the above that Jitsi handles sending RTP in changed
    codec,
    > but it does not work with received audio.
    >
    > As well: After change the codec, SIP Server does not change SSRC and
    > sends stream in new codec with old SSRC. maybe it is cause of the
    issue?
    >
    > Thanks!
    >
    > --------
    > Best regards,
    > Oleg

    --
    https://jitsi.org

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