On Thu, Oct 30, 2008 at 8:05 AM, Graham Wooden graham@g-rock.net wrote:
I using 1.4.0 with the new NAT Transversal module, and it so far it handles all my NATed clients; even folks that have devices that don't support STUN (like the older Polycom IP Soundpoint phones). So in this case, the above statement is not true with me as I am not proxing their audio.
I only proxy media under certain circumstances, like a court-ordered subpoena (CALEA), call re-direction support (which I haven't got fully working yet), or virtual fax and other media services (voicemail, conf calls, etc) from which the audio goes straight to my asterisk machines. And even with those, those are on a per-caller basis.
While this is starting to get off-topic, I have to ask:
Have you ever actually received a subpoena? Are you a CLEC? What is your interconnection to the PSTN?
The only reason I ask is because this sounds a little suspect. In most cases, telecoms CALEA is accomplished with LI capable software on various media devices and a third party subscription based service (like the one from Verisign) with direct or VPN access to twiddle the SNMP bits to achieve compatibility with standards like ATIS-1000678.2006. You can't just trap RTP... If you are an "interconnected VoIP provider" you have to provide full CALEA compliance to the relevant ATIS/TIA standards or figure out how you can get someone to do it for you. In many cases this can be easily provided by the small handful of multi-billion dollar orgs that provide these services in the US - Level(3), AT&T, Verizon Biz, XO, etc.
The only time I've ever been *aware* of a wiretap was when the customer authorized the monitoring: A couple of weeks ago a customer of ours was hosting an event for a current US Presidential candidate and the US Secret Service approached him asking for the contact information of his provider (us). The agent called me and faxed over the authorization, which I verified and forwarded. Other than that, I never hear about it...
With each g711u call leg, taking around 85kbps - that's 170 for each handled call ... 85 in, 85 out ... you can really start eating away at bandwidth.
Isn't your bandwidth symmetric/full duplex? How is 170kbps valid?
Plus, I am finding that the call quality is a bit better when the audio goes directly from the NAT client straight to the PSTN provider. While we do operate our own network (AS / BGP, with two Tier1 and Tier2 providers), if I don't have to proxy the audio, the better.
Totally makes sense in most cases:
- Depending on your connectivity - Depending on your SIP/PSTN provider - Depending on the customer's connectivity
..snip..