[asterisk-dev] A solution for including "RTP trunking" in Asterisk: Simplemux

[Jose Saldana]

Hi all,

 

I am Jose Saldana, a researcher from University of Zaragoza, in Spain. I am
new in this list!

 

In our research group we are working on a small-packet grouping solution,
which may be of interest for "call trunking" between two Asterisks.

 

I know that IAX2 supports trunking, but some Asterisk's users prefer SIP+RTP
for different reasons. Therefore, including an "RTP trunking" solution in
Asterisk could be an interesting feature. The name of our proposal is
Simplemux.

 

We have proposed it to the IETF, and we also have a running implementation.

 

The savings can be huge (up to 50% for certain voice codecs), as
multiplexing is combined with ROHC header compression.

 

If the Developers team find it interesting, I think it could be easily
integrated into Asterisk.

 

Some links:

 

A presentation:
http://es.slideshare.net/josemariasaldana/simplemux-traffic-optimization
(Asterisk scenario is in slide 12; results with VoIP are in slides 23-24)

 

The implementation in GitHub: https://github.com/TCM-TF/simplemux

 

The IETF draft:
https://datatracker.ietf.org/doc/draft-saldana-tsvwg-simplemux/

 

A couple of scientific papers explaining the idea:

http://diec.unizar.es/~jsaldana/personal/yoda_commag_2013.pdf

http://diec.unizar.es/~jsaldana/personal/chicago_CIT2015_in_proc.pdf

 

Best regards,

 

Jose Saldana, PhD
Dpt. Electrical Engineering and Communications
EINA, University of Zaragoza.
Ada Byron Building, D. 2.05
50018 Zaragoza, Spain

Tel: +34 976 76 2698

Ext: (84)2698

E-mail:  <mailto:jsaldana at unizar.es> jsaldana at unizar.es

 <http://diec.unizar.es/~jsaldana> http://diec.unizar.es/~jsaldana

 

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