Asterisk Playback sound dropping on linphone

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Asterisk Users 3 Comments

Hi,
I dial on A* from a linphonec to a Playback() extension, then suddenly
the sound stops after a while, without any notice.
I enabled debug both in linphone and A*, and the RTP packets are sent
from A* and received from linphone. It doesn’t matter whether I choose
alaw, ulaw, gsm as codec (besides changing cpu load of course).

How can I debug it? I’m using A* 1.6.2 and both linphone 2.x and 3.x.

I just need a console scriptable softphone, so maybe there’s an
alternative to linphone (which seemed good enough anyway!)…

Thank you,
Matteo

3 thoughts on - Asterisk Playback sound dropping on linphone

  • I did some more tests, and it’s not really a problem with linphone: the
    rtp capture shows empty packets sent by Asterisk.
    Since the channel which is doing Playback() is in a MeetMe conference, I
    tried also to speak on another phone on the same conference: well the
    rtp capture shows the stream from A* becoming silent, then the new sound
    from the phone comes up.

    Do I have to file a bug?

    Thank you,
    Matteo

    Il 11/11/2010 16:35, Matteo Fortini ha scritto:

  • Hi

    I use linphonec as well – and haven’t found another console sip phone
    either. I’d be interested if there is another one.

    Sebastian

  • There are some other clients, even if they are mainly testing/demo
    applications for some SIP stacks.

    sofsip-cli for SofiaSIP (which is backed by Nokia)
    simpleopal for OpalVOIP

    They do work, even if they’re not as full featured as linphone in some
    ways, e.g. on soundcard management. They offer some more options in
    other fields.

    Regarding the Playback issue, it seems that Playback into a
    [ConfBridge|MeetMe] conference stutters and drops randomly. I think I’ll
    file a bug for that.

    Thank you

    Il 12/11/2010 10:23, Sebastian ha scritto: