Without mediaproxy? Only based on SIP messages? 2008/10/29 Alex Balashov <abalashov(a)evaristesys.com>

Ok... Maybe you're right. I've read somewhere that this service is needed for taping reasons (policy and other law enforcements). If it's needed whe can just turn it on for that specific number, right? But answering to my question, can you point me some ideas refering about equipment that i should use? BR Nuno 2008/10/29 Alex Balashov <abalashov(a)evaristesys.com>

Hi everyone again, To make a point on this subject, and try to resume all that was said: 1º If i need a real VoIP system were billing is made by second (as it is in all the mobile providers) i need to have full control of the media - then it's necessary to implement mediaproxy (or other media relay) to accomplish accurate billing and get all the call information; 2º If the system doesn't need to have the accurate billing system, and doesn't matter if the call duration is 1 minute or 1,5 minute - for example a service that is paid monthly independently of the calls made by the user - i don't need to use mediaproxy in all calls, only in the NATed ones; I think that's what i can resume of all that was said. My problem now is to dimensioning this system to accomplish something like having 100 calls simultaneously (lets say all with mediaproxy - the "worst" case). Mark Sayer said that is system is based on Asterisk and he can accomplish 200 calls simultaneously. Can anyone give more ideas in this matter? Thanks and regards Nuno 2008/10/30 Henning Westerholt <henning.westerholt(a)1und1.de>

On 10/30/08 5:49 AM, "Nuno Marques" <yangsengopenser(a)gmail.com> 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 don¹t 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. With each g711u call leg, taking around 85kbps - that¹s 170 for each handled call ... 85 in, 85 out ... you can really start eating away at bandwidth. 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 don¹t have to proxy the audio, the better. And for the record, I have yet to come across a billing issue. I have clients that pay for unlimited service and I have ones that pay on bundled minutes plan. Hope this helps and regards, -graham

On Thu, Oct 30, 2008 at 8:05 AM, Graham Wooden <graham(a)g-rock.net> wrote: 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... Isn't your bandwidth symmetric/full duplex? How is 170kbps valid? Totally makes sense in most cases: - Depending on your connectivity - Depending on your SIP/PSTN provider - Depending on the customer's connectivity ..snip.. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

On Thu, Oct 30, 2008 at 9:25 AM, Graham Wooden <graham(a)g-rock.net> wrote: +1 to never being a CLEC... :) I bet if you were to look at those that have received subpoenas the commonality would be CLEC status + owning/operating the gateway. Yep, just making sure. Amen! If you are setting up media directly between your customers and your providers media gateways, how do you know what path it takes (in either direction)? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

The best part of all this is being able to say that the QoS isn't your problem and that the Internet sucks, right? :-) Graham Wooden wrote: -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

Jai Rangi wrote: Really? Next time I will consult with your authority on what works and does not work in production environments before implementing for large-scale billing solutions that are perfectly functional, and indeed, very much in production. By the way, there are, of course mitigating strategies to minimise risk. Dialog-stateful modules can end the dialog after a certain timeout, you can send periodic re-invites with an SDP offer to probe the endpoints, etc. It is far wiser than introducing a point of failure, a source of latency, and a source of huge bandwidth and processing cost into the call path when you don't need it. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

We are running a similar operation but settled on just OpenSER + Asterisk. All media runs thru Asterisk so MediaProxy isn't required and a custom billing engine was easy enough to put together. Yes it uses a lot more bandwidth and CPU but the combination of accurate accounting and easy NAT transversal make it worthwhile for us. The total number of subscribers isn't an issue (there are just entries in a database) there is no difference between VoIP<>VoIP and VoIP<>PSTN for us as both are SIP connections. We haven't maxed our initial Asterisk box yet but anticipate that it will handle about 200 concurrent calls depending on codec translation. A single OpenSER + database box should be able to handle several Asterisk boxes. (I'll also take comments on above, thanks) Mark At 10:56 a.m. 30/10/2008, you wrote: