Hi,
I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work with this requisites:
- 10000 users; - 100 VoIP to VoIP calls simultaneously capacity; - 30 VoIP to PSTN calls simultaneously capacity;
Can anyone point me some ideas of how can i design such a system (how many servers, how to distribute the services among them, etc.). I have this prototype mounted with VMWare, so i think that even making tests with sipp aren't going to be reliable. Thanks in advance,
Nuno
Nuno Marques wrote:
Every calls should pass through mediaproxy so that i can account them.
You can do accounting without handling media.
Without mediaproxy? Only based on SIP messages?
2008/10/29 Alex Balashov abalashov@evaristesys.com
Nuno Marques wrote:
Every calls should pass through mediaproxy so that i can account them.
You can do accounting without handling media.
-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
Yes. There are some liabilities with that in that the signaling messages may be incomplete (i.e. you may miss a BYE) and this is the usual reason given for doing media proxying for more accurate accounting.
But the latency, bandwidth consumption, and increased complexity and cost associated with doing it on a large scale does not justify it, in my opinion. SIP-only accounting is "good enough" most of the time.
Nuno Marques wrote:
Without mediaproxy? Only based on SIP messages?
2008/10/29 Alex Balashov <abalashov@evaristesys.com mailto:abalashov@evaristesys.com>
Nuno Marques wrote: Every calls should pass through mediaproxy so that i can account them. You can do accounting without handling media. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
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@evaristesys.com
Yes. There are some liabilities with that in that the signaling messages may be incomplete (i.e. you may miss a BYE) and this is the usual reason given for doing media proxying for more accurate accounting.
But the latency, bandwidth consumption, and increased complexity and cost associated with doing it on a large scale does not justify it, in my opinion. SIP-only accounting is "good enough" most of the time.
Nuno Marques wrote:
Without mediaproxy? Only based on SIP messages?
2008/10/29 Alex Balashov <abalashov@evaristesys.com mailto: abalashov@evaristesys.com>
Nuno Marques wrote:
Every calls should pass through mediaproxy so that i can account them.You can do accounting without handling media.
-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
On Thursday 30 October 2008, Nuno Marques wrote:
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?
Hi Nuno,
at least in some countries you must prevent the caller from noticing the surveilance. So if you transfer only some media of certain users this may lead to problems in this regards.
Cheers,
Henning
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@1und1.de
On Thursday 30 October 2008, Nuno Marques wrote:
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?
Hi Nuno,
at least in some countries you must prevent the caller from noticing the surveilance. So if you transfer only some media of certain users this may lead to problems in this regards.
Cheers,
Henning
Nuno Marques wrote:
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;
No, that's false.
Once again, you do not need to proxy media in order to do accurate call accounting with a 1 second granularity.
-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
On 10/30/08 5:49 AM, "Nuno Marques" yangsengopenser@gmail.com wrote:
...² i don't need to use mediaproxy in all calls, only in the NATed ones² ....
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
Graham Wooden wrote:
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.
Ditto.
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..
Quoting Kristian Kielhofner kkielhofner@star2star.com:
Have you ever actually received a subpoena? Are you a CLEC? What is your interconnection to the PSTN?
I have not directly received one, but know of folks that have. No, I am not a CLEC (nor plan to ever be one). My connection to the PSTN is SIP directly to the provider's SONUS switch.
Isn't your bandwidth symmetric/full duplex? How is 170kbps valid?
Client -->[~85kbps] -->device proxying audio -->[~85kbps] -->PSTN
If both of those legs come in and out on your same Internet provider leg, well, that call is going to cost you 170kbps. Since I do run BGP across multiple providers, I do have a fair bit of asymmetric routing, where the client may come in on my Tier2 and then sholve the call back out on my Tier1. Still adds up to 170kbps no matter how you slice it. But again, since I don't run full-time audio proxing, I don't have to worry about the bandwidth being absorbed like this anymore.
Why tack on another N amout of router hops and ms to the call if you don't need to?
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
In which case, all my customers are under 120ms (all broadband or higher and not all are local), pretty much all under 12 hops to me and to the PSTN. Works fine for me and I pay a fair bit of money to have my solid internet connections ;-)
Thanks,
-graham
On Thu, Oct 30, 2008 at 9:25 AM, Graham Wooden graham@g-rock.net wrote:
Quoting Kristian Kielhofner kkielhofner@star2star.com:
Have you ever actually received a subpoena? Are you a CLEC? What is your interconnection to the PSTN?
I have not directly received one, but know of folks that have. No, I am not a CLEC (nor plan to ever be one). My connection to the PSTN is SIP directly to the provider's SONUS switch.
+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.
Isn't your bandwidth symmetric/full duplex? How is 170kbps valid?
Client -->[~85kbps] -->device proxying audio -->[~85kbps] -->PSTN
If both of those legs come in and out on your same Internet provider leg, well, that call is going to cost you 170kbps. Since I do run BGP across multiple providers, I do have a fair bit of asymmetric routing, where the client may come in on my Tier2 and then sholve the call back out on my Tier1. Still adds up to 170kbps no matter how you slice it. But again, since I don't run full-time audio proxing, I don't have to worry about the bandwidth being absorbed like this anymore.
Yep, just making sure.
Why tack on another N amout of router hops and ms to the call if you don't need to?
Amen!
In which case, all my customers are under 120ms (all broadband or higher and not all are local), pretty much all under 12 hops to me and to the PSTN. Works fine for me and I pay a fair bit of money to have my solid internet connections ;-)
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)?
Quoting Kristian Kielhofner kkielhofner@star2star.com:
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)?
Good question; down here on the gulf coast, I know pretty much how the routes look to the PSTN from my current and future customers. Mainly because there isn't a bunch of broadband carriers to choose from. I know how all of them operate and how they are seen out on the 'net. However, my business has been growing and getting customers in other parts of the US and in those cases I just have them do a traceroute or mtr output. If I had a customer who did have a shaky connection to the PSTN but OK to me, in that case I would proxy the audio. But that would be a very rare case I would expect (haven't had to do that yet).
Thanks!
-graham
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:
Quoting Kristian Kielhofner kkielhofner@star2star.com:
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)?
Good question; down here on the gulf coast, I know pretty much how the routes look to the PSTN from my current and future customers. Mainly because there isn't a bunch of broadband carriers to choose from. I know how all of them operate and how they are seen out on the 'net. However, my business has been growing and getting customers in other parts of the US and in those cases I just have them do a traceroute or mtr output. If I had a customer who did have a shaky connection to the PSTN but OK to me, in that case I would proxy the audio. But that would be a very rare case I would expect (haven't had to do that yet).
Thanks!
-graham
Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
"SIP-only accounting is "good enough" most of the time." Does not work in production environment. Specially when you are charging per second or per minute. Works only if some one is offering unmetered only service or just doing it for fun. If it metered service like calling cards, termination or metered DID etc, then this can be really bad. My 2 cents.
-Jai "Buy unmetered SIP DID www.didforsale.com"
On Wed, Oct 29, 2008 at 3:56 PM, Alex Balashov abalashov@evaristesys.comwrote:
Yes. There are some liabilities with that in that the signaling messages may be incomplete (i.e. you may miss a BYE) and this is the usual reason given for doing media proxying for more accurate accounting.
But the latency, bandwidth consumption, and increased complexity and cost associated with doing it on a large scale does not justify it, in my opinion. SIP-only accounting is "good enough" most of the time.
Nuno Marques wrote:
Without mediaproxy? Only based on SIP messages?
2008/10/29 Alex Balashov <abalashov@evaristesys.com mailto:abalashov@evaristesys.com>
Nuno Marques wrote: Every calls should pass through mediaproxy so that i can account them. You can do accounting without handling media. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
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Jai Rangi wrote:
"SIP-only accounting is "good enough" most of the time."
Does not work in production environment.
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.
Hello, I can't say the " Works only if some one is offering unmetered only service or just doing it for fun. If it metered service like calling cards, termination or metered DID etc, then this can be* really bad*." If the Service providers use the Sip-B2BUA inside the Sip-Proxy servers. then, it sounds Good.
Thanks & Regards Ravi Prakash Sunkara VoIP Architect & JAVA-SIP Developer +91-9999882776
2008/10/30 Jai Rangi jprangi@gmail.com
"SIP-only accounting is "good enough" most of the time." Does not work in production environment. Specially when you are charging per second or per minute. Works only if some one is offering unmetered only service or just doing it for fun. If it metered service like calling cards, termination or metered DID etc, then this can be really bad. My 2 cents.
-Jai "Buy unmetered SIP DID www.didforsale.com"
On Wed, Oct 29, 2008 at 3:56 PM, Alex Balashov abalashov@evaristesys.comwrote:
Yes. There are some liabilities with that in that the signaling messages may be incomplete (i.e. you may miss a BYE) and this is the usual reason given for doing media proxying for more accurate accounting.
But the latency, bandwidth consumption, and increased complexity and cost associated with doing it on a large scale does not justify it, in my opinion. SIP-only accounting is "good enough" most of the time.
Nuno Marques wrote:
Without mediaproxy? Only based on SIP messages?
2008/10/29 Alex Balashov <abalashov@evaristesys.com mailto:abalashov@evaristesys.com>
Nuno Marques wrote: Every calls should pass through mediaproxy so that i can account them. You can do accounting without handling media. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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:
Yes. There are some liabilities with that in that the signaling messages may be incomplete (i.e. you may miss a BYE) and this is the usual reason given for doing media proxying for more accurate accounting.
But the latency, bandwidth consumption, and increased complexity and cost associated with doing it on a large scale does not justify it, in my opinion. SIP-only accounting is "good enough" most of the time.
Nuno Marques wrote:
Without mediaproxy? Only based on SIP messages?
2008/10/29 Alex Balashov <abalashov@evaristesys.com mailto:abalashov@evaristesys.com>
Nuno Marques wrote: Every calls should pass through mediaproxy so that i can account them. You can do accounting without handling media. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
Thanks Mark for your answer! If you're using only OpenSER and Asterisk, this means that you pass all your calls through Asterisk and use openser only for authentication and proxying. Is that right? How do you make the calls pass through in NAT environments? Is Asterisk capable of handling that? How?
Too many questions... i know... but i'm excited with that possibility...
BR
Nuno
2008/10/30 Mark Sayer datapipes@avtb.co.nz
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:
Yes. There are some liabilities with that in that the signaling messages may be incomplete (i.e. you may miss a BYE) and this is the usual reason given for doing media proxying for more accurate accounting.
But the latency, bandwidth consumption, and increased complexity and cost associated with doing it on a large scale does not justify it, in my opinion. SIP-only accounting is "good enough" most of the time.
Nuno Marques wrote:
Without mediaproxy? Only based on SIP messages?
2008/10/29 Alex Balashov <abalashov@evaristesys.com mailto:abalashov@evaristesys.com>
Nuno Marques wrote: Every calls should pass through mediaproxy so that i can account them. You can do accounting without handling media. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
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Alex Balashov -
Graham Wooden -
Henning Westerholt -
Jai Rangi -
Kristian Kielhofner -
LetMeKnow -
Mark Sayer -
Nuno Marques