It depends on how you did your tests..That's weird. I've done some testing both with GS and Xten products, and my iptraf readings show much more than your numbers.
If you ran iptraf an PC1 and made a call from PC2(X-Lite) to GS and your sip.conf entry for either have canreinvite=no then you will get double the traffic..
Best bet is to run iptraf on the Asterisk box and then make a call from the phone to Asterisk (eg to voicemail, echo test, the pstn or a Zap channel) so that the IP traffic is only one client making a call to Asterisk using the selected codec.. That should give you the best reading..
Later..
Paulo Mannheimer wrote:
Hi All-
I'm working on a project that will have remote (internet)access to an *
server through SIP phones, either soft or hard ones.Depends on the phone.. If you are using a Grand Stream then the best you
Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage?
TIA!
PauloHM
will get is G.711 (+- 85Kb/s including overheads)..
If you are using Snom's or X-Lite/X-Pro you have the option to use the GSM (+- 34Kb/s including overheads) codec..
X-Lite/X-Pro also support iLBC (+- 28Kb/s including overheads) although it does not currently work with Asterisk, and GrandStream have said they
are going to support it as well soon..
All the phones have support for G.729 (+- 22Kb/s) either as standard or by buying a sepertate licence.. Including Asterisk..
Hope that helps..
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