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

[Adrian Musceac]

Hi Albin,

Sorry, I was under the impression that RTP would be transmitted over
the air, which in some cases might make sense... like IP multicast
over MPEG-TS over DVB-S. But for Codec2 it does seem like raw audio
transmission is better suited. So then why the need for a Codec2/Opus
RTP block? I would think it would be better left to the application
layer outside GNU Radio to handle IP routing of audio streams?

My previous approach was to use GNU Radio blocks only for the PHY
layer, and use a simple and robust VOIP protocol (Mumble) to
distribute audio over TCP/IP, with some transcoding involved in case
of FM and Codec2 digital voice. With a single GNU Radio instance I
could then serve up to 4 voice channels and one control channel using
FDMA, with a mix of analog and digital channels.
Then again, this is just an amateur radio perspective, where network
capacity is not such a big issue and infrastructure density or SNR
performance are the limiting factor.

Regards,
Adrian YO8RZZ

On 5/10/18, Albin Stigö <address@hidden> wrote:
> 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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>