Pause-Resume with Speex RTP Source

[Scott Kidder]

I'm using the Speex audio codec over RTP to deliver near-realtime audio and
have had good results except under the following condition.  I'm using the
iOS GStreamer development library with the following pipeline:

Client:
udpsrc port=16500 ! application/x-rtp, media=(string)audio,
clock-rate=(int)8000, encoding-name=(string)SPEEX,
encoding-params=(string)1, payload=(int)110 ! rtpspeexdepay ! speexdec !
audioconvert ! autoaudiosink


Server:
jackaudiosrc ! audioconvert ! audioresample !
'audio/x-raw-int,rate=8000,width=16,channels=1' ! speexenc ! rtpspeexpay !
udpsink host=224.0.0.1 port=16500


I modified the iOS GStreamer 'Tutorial 2' app, which uses the following
function calls to pause & play the stream:

    [gst_backend pause];

    [gst_backend play];

The problem is that the client application buffers the audio while paused.
 I'd like the client application to discard all packets received while
paused, and resume from the current timestamp.

This worked when using raw PCM audio, but does not work with the Speex
codec.  Are there any suggestions for how to change the pipeline and/or
application?

Thanks,

--Scott Kidder
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