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.
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