RTP server and client

[aft]

On Tue, Feb 11, 2014 at 1:19 AM, Sebastian Dröge
<sebastian at centricular.com> wrote:
> On Mo, 2014-02-10 at 13:20 +0600, aft wrote:
>> Hi,
>>
>> I'm trying to tryout out the following two pipelines :
>>
>> [1] RTP server :
>>
>>       $ gst-launch-0.10 -v gstrtpbin name=rtpbin \
>>        autiotestsrc ! audioconvert !alawenc !rtppcmapay !
>> rtpbin.send_rtp_sink_0  \
>>              rtpbin.send_rtp_src_0 ! udpsink port=5002 host=127.0.0.1
>>
>> [2] RTP client :
>>   gst-launch-0.10 -v gstrtpbin name=rtpbin
>>                    \
>>             udpsrc
>> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"
>> port=5002 ! rtpbin.recv_rtp_sink_0
>>  rtpbin.     ! rtppcmadepay ! alawdec !audioconvert ! audioresample
>> !autoaudiosink
>>
>>  Both pipeline are running, but nothing is happened, i can't hear the
>> "beep" tone generated from audiotestsrc.
>
> First of all try upgrading to GStreamer 1.2.x to see if your problem
> disappears then. There were many bugfixes around this.
>
> If that doesn't help try setting multicast-group=127.0.0.1 on your
> client. And check with tools like wireshark if anything is sent at all
> and to which IP.
>

Thanks for the tip. Its working with 1.2.3....

>
> Also it would probably make sense to also use the RTCP parts of rtpbin
> on both the sender and receiver.
>
> --
> Sebastian Dröge, Centricular Ltd - http://www.centricular.com
> Expertise, Straight from the Source
>
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>



-- 
-Cheers
-Arif