Hello Maxim,
Monday, March 7, 2005, 12:58:03 AM, you wrote:
MS> Alistair Cunningham wrote:
Matt,
Because SER handles SIP messages, not RTP streams, it cannot 100% accurately record CDRs. For instance, if the BYE message is lost because a client crashed, SER has no way of knowing when the call ended or how much it cost. A BTBUA gets round this problem by having the RTP packets come to it, so if a client crashes, the RTP stream stops and the BTBUA knows the call has finished. The disadvantage of a BTBUA is that it adds latency and doesn't scale well, because it needs to handle all the RTP traffic.
MS> This is not necessarily true. SIP permits building signalling-only MS> B2BUA, which if designed correctly will be able to handle up to 50-100 MS> call setups/teardowns per second on any modern x86 hardware. This will MS> provide capacity sufficient for any mid-level ITSP - allowing to bill MS> more than 10 millions per month. If this is not sufficient then several MS> of such B2BUA can be installed in parallel to provide more scalable MS> solution. The problem with using either Cisco or Asterisk as SIP B2BUA MS> is that they handle both signalling (SIP) and media (RTP), which makes MS> them less scalable and introduces the problem with increased latency, MS> losses and jitter.
MS> Regards,
MS> Maxim
I install quite a few SER and Asterisk systems for my customers, and a fix for this issue is the most frequent item I'm asked for. It's unavoidable, however, if one wants SER's scalability. What I usually recommend for them is to install a SER system (for scalability, registrations, routing of calls that don't need a CDR, etc) fronting multiple BTBUAs such as Asterisk or Cisco (for accurate billing of PSTN calls).
Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/
matt morris wrote:
Hello List,
Just wondering, what is the radius account module for SER not able to do, that would require the use of B2BUA in terms of having a CDR? Forgive me if it is a dumb question...
Thanks.
I will not agree that asterisk proxy RTP streams always, you can set asterisk not to do it. But will agree that asterisk is not developed for such purposes and some limitations exists (but some additional features exists too).