[gstreamer-bugs] [Bug 613248] Interop: no mic when calling Polycom Soundpoint phone.

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=613248
  GStreamer | gst-plugins-base | git

Sjoerd Simons <sjoerd> changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
                 CC|                            |sjoerd at luon.net
          Component|VoIP                        |gst-plugins-base
            Version|2.29.x                      |git
         AssignedTo|empathy-maint at gnome.bugs    |gstreamer-bugs at lists.source
                   |                            |forge.net
            Product|empathy                     |GStreamer
   Target Milestone|---                         |HEAD
          QAContact|                            |gstreamer-bugs at lists.source
                   |                            |forge.net

--- Comment #3 from Sjoerd Simons <sjoerd at luon.net> 2010-04-08 09:56:26 UTC ---
Turns out that these polycom phones only work if every RTP packets contains a
multiple of 10ms of audio, anything else breaks. Unfortunately there is
currently no way to indicate this requirements to the gstreamer rtp payloaders.
TBH i wouldn't be surprised if more phones acted in this way, so it might not
be a bad idea to by default use a multiple of 10ms of audio (at least for
PCMU/PCMA).

Olivier should have a proof of concept patch for this which we used to test
against the polycom phone

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