RTP retransmission mechanism in gstreamer 1.2.0

[George Kiagiadakis]

Hi,

On Wed, Dec 11, 2013 at 11:15 AM, abhilash <abhilashtuse at gmail.com> wrote:
> Hi,
>
> Thanks for your reply.
> As you suggested I tried using rtprtxqueue on receiver side(pipeline below):
>
> gst-launch-1.0 -v rtpbin name=rtpbin udpsrc port=5000
> caps="application/x-rtp, media=audio, clock-rate=8000, encoding-name=PCMA,
> payload=8" ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtprtxqueue ! rtppcmadepay !
> alawdec !
>
> alsasink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0  rtpbin.send_rtcp_src_0
> ! udpsink host=192.154.10.21  port=5005 sync=false async=false
> --gst-debug=3
>
> Still I am not able to see any RTCP packets other than reports from receiver
> to sender in wireshark.
> Is above pipeline correct?
> As per my understanding there should be RTCP request from receiver to sender
> to send again lost packets.
> Is this correct?
>

You misunderstood what I said. I said you need to use the
do-retransmission=true on the receiver side and that rtprtxqueue needs
to be added after the *pay*loader, which lives on the sender side.

Regards,
George

On Wed, Dec 11, 2013 at 11:15 AM, abhilash <abhilashtuse at gmail.com> wrote:
> Hi,
>
> Thanks for your reply.
> As you suggested I tried using rtprtxqueue on receiver side(pipeline below):
>
> gst-launch-1.0 -v rtpbin name=rtpbin udpsrc port=5000
> caps="application/x-rtp, media=audio, clock-rate=8000, encoding-name=PCMA,
> payload=8" ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtprtxqueue ! rtppcmadepay !
> alawdec !
>
> alsasink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0  rtpbin.send_rtcp_src_0
> ! udpsink host=192.154.10.21  port=5005 sync=false async=false
> --gst-debug=3
>
> Still I am not able to see any RTCP packets other than reports from receiver
> to sender in wireshark.
> Is above pipeline correct?
> As per my understanding there should be RTCP request from receiver to sender
> to send again lost packets.
> Is this correct?
>
> Thanks,
> Abhilash.
>
>
>
> On Wed, Dec 11, 2013 at 3:21 PM, George Kiagiadakis [via GStreamer-devel]
> <[hidden email]> wrote:
>>
>> On Wed, Dec 11, 2013 at 10:29 AM, abhilash <[hidden email]> wrote:
>>
>> > Hi,
>> >
>> >      I am trying out a client-server based audio streaming application
>> > using
>> > gstreamer 1.2.0 version, running on 2 different Linux machines on
>> > network. I
>> > want to check the functionality of retransmissions as mentioned in the
>> > "rtpbin" element and I tried setting the property
>> > "do-retransmissions=true".
>> > I am simulating packet loss, packet delay & reorder on the server side.
>> > The
>> > observation is as follows -
>> > 1.  I am able to see RTCP Sender and Receiver reports being sent between
>> > server and client on Wireshark.
>> > 2.  I am able to see dropped RTP packet sequence numbers in Wireshark
>> > trace,
>> > however I am not able to see any RTP retransmissions event going from
>> > client
>> > to server.
>> >
>> >     The pipelines at the server and client side are as mentioned below -
>> >
>> > Server:
>> > gst-launch-1.0 -v rtpbin name=rtpbin do-retransmission=true audiotestsrc
>> > freq=1000 ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0
>> > rtpbin.send_rtp_src_0 ! udpsink host=192.154.10.20 port=5000
>> > rtpbin.send_rtcp_src_0 ! udpsink host=192.154.10.20 port=5001 sync=false
>> > async=false udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 --gst-debug=3
>> >
>> > Client:
>> > gst-launch-1.0 -v rtpbin name=rtpbin udpsrc port=5000
>> > caps="application/x-rtp, media=audio, clock-rate=8000,
>> > encoding-name=PCMA,
>> > payload=8" ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtppcmadepay ! alawdec !
>> > alsasink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0
>> > rtpbin.send_rtcp_src_0
>> > ! udpsink host=192.154.10.21  port=5005 sync=false async=false
>> > --gst-debug=3
>> >
>> >      Can anyone please let me know if anything is missed out above and
>> > why I
>> > am not seeing any RTP retransmissions ?
>> >
>> Hi,
>>
>> do-retransmission=true should be set on the receiver (client) rtpbin.
>> On the sender it has no effect. Also, since you are on 1.2, you need
>> to add rtprtxqueue in your pipeline after the payloader. Note though
>> that this does not implement RFC4588-compliant retransmission packets.
>> An RFC4588-compliant implementation is under review in bugzilla at the
>> moment.
>>
>> Regards,
>> George
>> _______________________________________________
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>
>
>
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> 1.2.0
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