[asterisk-biz] How Realistic is Hosted VoIP for SMBs?

[Leo Ann Boon]

GlobalOfficePhone wrote:
> Greg:
>
> Thanks for your thoughts. I am familiar with SIP signalling and media 
> stream separation in SIP and I agree with you that the media can be 
> sent directly between two SIP endpoints. This would work if the PBX 
> was local and the PBX and the endpoints were behind the same NAT.

> However, as I understand it,  when the endpoints are behind a 
> different NAT then the hosted PBX (which most are in case of SMBs), 
> then SIP and Asterisk doesn't allow direct endpoint-to-endpoint 
> connection and the media stream must pass through the Asterisk server. 
> Unless, of course, the hosted PBX is using some other tricks.
Direct RTP between phones is still possible for a remotely hosted PBX. 
Most hosted PBX solutions use a variety of tricks to optimize the traffic.

a. Deploy SIP aware router or outbound proxy at CPE end. In such a 
setup, the router or OP will rewrite the SIP/SDP such that all phones at 
the same site send RTP directly to each other.

b. In a multi-site setup, it's possible to have direct media if all the 
sites are using SIP aware router and outbound proxy.

c. For isolated remote phones (no SIP aware router/OP), the phones can 
always try STUN, UPNP first before resorting to using hosted SBC 
(Session Border Controller).  Google Kagoor or Acmepacket for examples 
of SBC.
>
> As per your last comment, my only point is that if 
> endpoint-to-endpoint media stream is not possible, then the situation 
> described would require much higher WAN bandwidth.
Yes, if using an SBC to proxy/relay the calls.

Leo.