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Re: [Discuss-gnuradio] RTP block / Opus vocoder


From: Albin Stigö
Subject: Re: [Discuss-gnuradio] RTP block / Opus vocoder
Date: Thu, 10 May 2018 22:49:40 +0200

Hi Adrian,

UDP and RTP adds a lot of overhead to a codec like Codec2 and doesn't
make any sense at all unless you wan't to route your packets over an
IP WAN like the internet. Then it makes a lot of sense.

I imagine the only use case for an RTP/Codec2 or RTP/Opus block is
streaming audio from a remote receiver/transceiver over an existing IP
network. If you only stream over LAN, UDP is good enough but the
Internet an stack RTP is better. All streaming software like skype,
facetime, hangouts etc use RTP.


--Albin SM6WJM

On Thu, May 10, 2018 at 10:05 PM, Adrian Musceac <address@hidden> wrote:
>>
>> We would like to combine Opus/CODEC2 and RTP multicast to have stereo field
>> audio. The sources of the audio appear at different points in the stereo
>> field, so that a roundtable conversation feels more like a roundtable, or
>> so that two streams from two different SDRs are distinct.
>>
>
> Hi Michelle,
>
> This is very interesting to me. I did some work with Codec2, Opus and
> GNU Radio myself.
> I'm curious though: RTP and UDP + IP encapsulation, isn't that a bit
> too much overhead for a low bitrate codec like Codec2? A 40 msec audio
> frame encoded with Codec2 1400 is only 48 bits, so the encapsulation
> overhead is quite large. How does that affect low SNR performance?
>
> Regards,
> Adrian YO8RZZ
>
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