Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
Without mediaproxy? Only based on SIP messages? 2008/10/29 Alex Balashov [EMAIL PROTECTED] 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
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 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
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 [EMAIL PROTECTED]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 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
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. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
Really? Yes, Specially when your service is metered, I don't know how some once justify good enough billing. Dealing with 500 customer calling every day for billing inquiries can turn out to be much more expensive then all other expenses. 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. No Need to be so contemptuous. On Wed, Oct 29, 2008 at 4:37 PM, Alex Balashov [EMAIL PROTECTED]wrote: 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. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
By good enough I really did mean good enough, not sort-of kind-of okay. Jai Rangi wrote: Really? Yes, Specially when your service is metered, I don't know how some once justify good enough billing. Dealing with 500 customer calling every day for billing inquiries can turn out to be much more expensive then all other expenses. 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. No Need to be so contemptuous. On Wed, Oct 29, 2008 at 4:37 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: 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. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
