6 Mar
2005
6 Mar
'05
9:38 p.m.
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.
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6 Mar
6 Mar
11:06 p.m.
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.
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.
_________________________________________________________________
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outbound e-mail and attachments.
http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&D…
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11:58 p.m.
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.
This is not necessarily true. SIP permits building signalling-only
B2BUA, which if designed correctly will be able to handle up to 50-100
call setups/teardowns per second on any modern x86 hardware. This will
provide capacity sufficient for any mid-level ITSP - allowing to bill
more than 10 millions per month. If this is not sufficient then several
of such B2BUA can be installed in parallel to provide more scalable
solution. The problem with using either Cisco or Asterisk as SIP B2BUA
is that they handle both signalling (SIP) and media (RTP), which makes
them less scalable and introduces the problem with increased latency,
losses and jitter.
Regards,
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.
_________________________________________________________________
Scan and help eliminate destructive viruses from your inbound and
outbound e-mail and attachments.
http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&D…
Start enjoying all the benefits of MSN® Premium right now and get the
first two months FREE*.
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7 Mar
7 Mar
12:20 a.m.
Maxim Sobolev wrote:
Alistair Cunningham wrote:
Maxim,
You're quite correct. I was generalising too much.
--
Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
http://integrics.com/
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.
This is not necessarily true. SIP permits building signalling-only
B2BUA, which if designed correctly will be able to handle up to 50-100
call setups/teardowns per second on any modern x86 hardware. This will
provide capacity sufficient for any mid-level ITSP - allowing to bill
more than 10 millions per month. If this is not sufficient then several
of such B2BUA can be installed in parallel to provide more scalable
solution. The problem with using either Cisco or Asterisk as SIP B2BUA
is that they handle both signalling (SIP) and media (RTP), which makes
them less scalable and introduces the problem with increased latency,
losses and jitter. 
9:35 a.m.
New subject: Re[2]: [Serusers] SER RADIUS Accounting and B2BUA
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).
--
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YES ISP, fx:380-3433-47067
Valova 17, mike|a|yes.net.ua
Kolomyia, www.yes.net.ua
Ukraine 78200 FWD: 66518
07.03.2005
ICQ# 57698805
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1:40 p.m.
At 11:06 PM 3/6/2005, 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.
It can be fairly accurate. You just need to provision the network carefuly and use
advanced
features such as session timer.
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.
Indeed. That's the pain. That's why B2BUA with media relay is to be used as
rarely as possible.
-jiri
3:01 p.m.
New subject: [Serusers] SER MESSAGES
Hi all.
Just two questions:
1.- I think that SER does not support presence for Windows Messenger 5.0
or above. Is that correct ? Is there anyway of supporting it (new PA
module, ...) ?
2.- Is there anyway of use instant messaging for Windows Messenger with
SER ? Which are the modules that I have to use ? When I try to log a
MESSAGE method [ with:
"if(method=="MESSAGE")
{
log(1, "MESSAGE");
}
I get nothing because first it tries with an INVITE packet and RTPROXY
takes it and sais that cannot extract media because I think he expects
audio or video media but no messages. Is there a way of do it?
Many thanks in advance.
4:58 p.m.
New subject: [Serusers] SER MESSAGES
Hi Kiko!
You can force Windows Messenger to fallback from "session model" to
"page model" by catching the INVITEs in your ser.cfg and reply with 488
- e.g.:
if ( method=="INVITE" && ( search("^m=x-ms-message")
||
search("^m=message") ) && !search("^m=audio") ) {
xlog("L_INFO", "%is [%Tf]: %rm %fu -> %ru [R4]: INVITE
with message session found -> reply 488!");
sl_send_reply("488", "Not Acceptable Here");
break;
}
regards,
klaus
Kiko Vives wrote:
Hi all.
Just two questions:
1.- I think that SER does not support presence for Windows Messenger 5.0
or above. Is that correct ? Is there anyway of supporting it (new PA
module, ...) ?
2.- Is there anyway of use instant messaging for Windows Messenger with
SER ? Which are the modules that I have to use ? When I try to log a
MESSAGE method [ with:
"if(method=="MESSAGE")
{
log(1, "MESSAGE");
}
I get nothing because first it tries with an INVITE packet and RTPROXY
takes it and sais that cannot extract media because I think he expects
audio or video media but no messages. Is there a way of do it?
Many thanks in advance.
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7210
days inactive
7211
days old
7 comments
7 participants
participants (7)
-
Alistair Cunningham -
Jiri Kuthan -
Kiko Vives -
Klaus Darilion -
matt morris -
Maxim Sobolev -
Mike Tkachuk