[Asterisk-Users] Sip bandwidth usage

[WipeOut]

Paulo Mannheimer wrote:

>That's weird. I've done some testing both with GS and Xten products, and
>my iptraf readings show much more than your numbers.
>
It depends on how you did your tests..

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 *
>>    
>>
>
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>>server through SIP phones, either soft or hard ones.
>>
>>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
>>
>> 
>>
>>    
>>
>Depends on the phone.. If you are using a Grand Stream then the best you
>
>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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