music on hold server using pre-encoded audio

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On Tue, Sep 23, 2008 at 8:08 AM, Andrew Radke <andrew.radke at yuruga.com.au>wrote:

> Hi,
> I hope this is a relatively simple question but I seem to be chasing my
> tail around the Internet looking for a solution.
>
>
The question is always simple (but the answer most likely is not). :)


> I need a music on hold service that supports G711 and G729 at a minimum. I
> have converted the wav file to ulaw using sox and then to G729 using this
> tool: http://www.asteriskguru.com/tools/audio_conversion.php
>
> The theory is that having it pre-encoded removes any requirement to
> actually understand the codec you are streaming and reduces the resource
> requirements to a minimum.
>
>
So if I read this correctly, you wanted to stream encoded audio from a file
to remote calls.


> I'm hoping that this would be a very simple thing to do with PJSIP/PJMEDIA
> since it already has support for dealing with RTP whereas most other
> frameworks I've found only do SIP, or where they do the audio too they need
> to understand or transcode it. I would also hope to end up with a simple set
> of files that can be copied onto a target system without a huge range of
> libraries, etc being required which also makes PJSIP look promising.
>
> If this isn't something that can be done with PJSIP a pointer in right
> direction would also be heartily appreciated.
>
>
Of course it can be done. Do you not think PJSIP as a magic library that can
do everything? ;-)

On a more serious note, pjmedia as a media library works mostly with PCM
frames and not encoded frames, for obvious reason. The RTP streaming is done
only at the edge of the processing where the frames are sent/received
to/from the transport/network, and this is done by an object called stream
(stream.c), where it does the conversion between PCM frames and RTP packets.
This includes encoding/decoding the audio, managing the RTP/RTCP session,
and de-jitter-buffering on the RX side.

So if you were to stream PCM WAV files, then it would have been trivial to
do, even the sample pjsua application can do this out of the box. But of
course the implication on the resources used and codec licensing (for g729)
would not be so trivial.

Having said that, it should be possible to stream an encoded file to the
network; all one has to do is to create something like stream, but instead
of taking PCM audio as the input, it takes encoded audio instead. And the
rest such as the RTP/RTCP session management and de-jitter-buffering would
need to be reimplemented on this object in similar ways like it's done in
stream.c or siprtp.c from the sample. This might sound like a daunting task,
but actually it shouldn't be that difficult, since all the lower level
components are there already, so it's just a matter of integrating them into
a new higher-level object. And it has been done by other people too.

Or alternatively, you can also hack stream.c so that it works with encoded
audio. We might do this sometime in the future since some audio devices
(such as APS on Symbian) are capable of doing hardware encoding, but
considering other stuffs that we need to do we don't have concrete plan to
do this yet.

So there you go, the options are yours. :)

Cheers
 Benny



> Regards,
> Andrew Radke
> Yuruga Nursery Pty Ltd
> Clonal Solutions Australia Pty Ltd
> PO Box 220
> Walkamin Qld 4872
> Phone: (07) 4093 3826
> Fax: (07) 4093 3869
> Email: andrew.radke at yuruga.com.au
> Web: www.yuruga.com.au
>
>
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