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Re: [Linphone-developers] Inhibit RTP transmission when mute is on in ac

From: barry krofchick
Subject: Re: [Linphone-developers] Inhibit RTP transmission when mute is on in active call
Date: Mon, 17 May 2010 20:25:20 -0400

Sven made a good point.

I have included a slightly more complex patch which makes RTP transmission on
audio mute fully configurable. This patch has been tested and works in linphonec,
linphonecsh and linphone-3 (gtk).

The key changes are as follows:

1. A new parameter in the rtp section of lpconfig was added:
        rtp_no_xmit_on_audio_mute (which has a default of 0 or off)
        If this parameter is set to 1 then rtp transmission will be off
        when audio is muted. Since the default is off, if this parameter
        is not specified in the config file all will function as it
        does now (rtp transmission continues when audio is muted)
 2. 3 new commands were added to commands.c (so that audio
     muting can be accessed from linphonec, and linphonecsh (as well
     as the linphone-3 gtk.
            1. mute (mutes audio and possibly rtp)
            2. unmute (unmutes audio and possibly starts rtp of it was off)
            3. nortp-on-audio-mute [1|0]
                   turns rtp muting on/off when audio muting on/off
                   (this is needed to make rtp muting work in linphonecsh mode
                    where there is no config file accessed)
    3. the command 'status hook' was modified to report the status
        of both audio and rtp muting (so that scripts may adjust if needed
        to these values)

   Thus to enable rtp muting (when audio muted) simply add
    to the rtp section of config file.

    To disable rtp muting leave the rtp_no_xmit_on_audio_mute
    out of the config file or set its value to 0.

    When using linphonecsh, since there is no config file associated with
     a call, if one wishes to enable rtp muting - all that is needed is to
     send the generic command:
             linphonecsh generic "nortp-on-audio-mute 1"

    The full patch with configuration, is attached below.

    Again , this was fully tested in all 3 modes.

    I hope that this can be merged into the mainline.

    In a steady state connection (using speex wb) if rtp muting is
    linked to audio muting , transmission bandwidth when audio is
    muted drops from about 6500 kbytes./sec to around 145 bytes/sec.

    Thus one can see that maintaining a SIP connection, for long periods
    of time, when connected to an automated backend agent, will use
    negligible bandwidth or server resources.

     Again linphone is great, the perfect SIP client for us with this little mod.

     The patch is in the attached file.



On Mon, May 17, 2010 at 2:01 AM, Sven Westergren <address@hidden> wrote:

Hi all,


I would ask that if this patch is merged it is configurable (at least at compile time) since some SIP clients will terminate the call if the media stream goes dead for a period of time.


Best regards

Sven Westergren


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