[asterisk-users] Re: VOIP Bandwidth questions

[Julio Arruda]

Eric "ManxPower" Wieling wrote:
> Vikki wrote:
>> I think vonage is using g723.1 which requires 6.4kbps voice bandwidth
>> compared to g711 - 64kbps.
>>
>> For SIP to SIP calls, RTP doesn't necessarily goes thru the server. Only
>> Signalling goes to the servers. This means no bandwidht usage for the
>> provider.
>> For SIP to PSTN calls, it has to goes thru a media gateway (owned by the
>> provider) which may be seperate from the sip server.
> 
> I imagine that most of Vonage's customers are behind NAT and direct RTP 
> (re-invites) don't work well with the endpoints behind NAT.

At least some solutions for 'remote end NAT'(not 'CPE solved the 
problem' NAT) can involve 'distributed' RTP Proxies, to anchor the RTP 
call legs.