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 To :  yate@v...
 From :  "Raffaele P. Guidi" <raffaele.p.guidi@g...>
 Subject :  Reducing bandwidth usage
 Date :  Tue, 17 Feb 2009 23:35:43 +0100
I'm using the GSM codec to connect to a sip provider. I soon found out that
while in literature gsm bandwidth usage is 13kbps the udp/sip wrapping
protocols bring that to 56kbs (a 4x increment!). Googling around I found
this post
http://lists.digium.com/pipermail/asterisk-users/2004-September/054167.html

in
the asterisk mailing list that mentions IAX trunking as a possible solution
to decrease bandwidth usage. Is that possible with yate, too? And is there
some kind of trunking available for the sip protocol?
Thanks,
   Raffaele



I'm using the GSM codec to connect to a sip provider. I soon found out that while in literature gsm bandwidth usage is 13kbps the udp/sip wrapping protocols bring that to 56kbs (a 4x increment!). Googling around I found this post http://lists.digium.com/pipermail/asterisk-users/2004-September/054167.html  in the asterisk mailing list that mentions IAX trunking as a possible solution to decrease bandwidth usage. Is that possible with yate, too? And is there some kind of trunking available for the sip protocol?

Thanks,
   Raffaele