Disturbed VOIP payload

[symeon.mattes]

Hi all,

I'm using streamer for VOIP payload and for some reason I have in the
receiving pipeline a disturbed payload. I have checked through wireshark the
network traffic and my voice seems to reach my device with no lost VOIP
packages.

So it seems that there is something on the gstreamer pipeline. My pipeline
is:

udpsrc->queue->rtpjitterbuffer->rtppcmadepay->alawdec->audioconvert->audioresample->queue->osxaudiosink

The udpsrc has a properties
    caps = gst_caps_new_simple ("application/x-rtp",
                                "media", G_TYPE_STRING, "audio",
                                "clock-rate", G_TYPE_INT, 8000,
                                "encoding-name", G_TYPE_STRING, "PCMA",
                                "payload", G_TYPE_INT, 8,NULL);


I tried to increase the max-size-time of the queue, the latency of the
rtpjitterbuffer but with no much success. There was much improvement by
removing totally from the pipeline the queue and the rtpjitterbuffer, so I
had

udpsrc->rtppcmadepay->alawdec->audioconvert->audioresample->osxaudiosink

But I still had some corruption on the voice. Could you give me any hints?
Is there any tool debugging tool that I could use in order to see what's
going wrong on the pipeline?


Thanks



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