Mailing List Archive

AS5350XM in mixed VoIP and dialup environment
Hi everyone,

I sent some of those questions to cisco-nsp already, but maybe this (or
at least for some parts cisco-voip) is the better forum.

we're currently evaluating two Cisco AS5350XM for use in our university
network. It should replace the old Ascend TNT boxes for ISDN/modem
dialup and provide a SIP/PSTN gateway to use for our VoIP PBX to be
installed next year. The whole setup will basically look like this

+----------------+
| PSTN |
+----------------+
| | | | | |
| | | | | | 6*E1
| | | | | |
+----------------+ E1 +-------------+
| AS5350XM |-----------| channel box |
+----------------+ +-------------+
+
+ SIP
+
+----------------+
| SIP PBX (*) |
+----------------+

Current setup is just one E1 to our HiCom.

All six E1 lines will be configured the same way and will have a large
block as well as several additional numbers configured (so calls to one
number can be signalled on one random E1).

Data (ISDN) or Voice (modem) calls to several numbers on that trunk
should be handled by the box itself (ordinary PPP dialup). A small block
of our numbers should be sent to the channel box (so that is basically
PSTN-to-PSTN switching). An important thing here would be that PSTN to
the channel box is transparent regarding data, so we can connect any
device there. All remaining destinations should be sent to the PBX with
SIP.

My first question is regarding dialup. Currently we have the whole
dialup configuration on Serial3/0:15 and additionally on Group-Async0,
which has both 108port spes configured in (group-range 1/00 2/107). This
makes the Cisco answer each and every call it receives with PPP. If I
wanted to connect to different "configuration profiles" depending on the
dialed number, I had to put a "dialer pool-member x" on the lines and
use it in several Dialer-interfaces with the same "dialer pool x" and
different "dialer called <number>", correct? Would that work with async
(modem) connections as well? As far as I understood the documentation it
won't, and I would need to use resource-pooling to assign specific
numbers to specific async lines and then group those lines specifically.

The second question is about the use of the SPE dsps. We patched some
lesser used lines/numbers to the two Ciscos we got and the output of "sh
spe" does not look to good. E.g. at the moment we have 5 modem users (no
compression on the group-async and no multilink) and 6 digital users
(multilink and compression allowed), of which three are ordinary
dialups, two have multilink enabled but only one channel in use and one
has two channels. Still, show spe shows

Ports : Total 216 In-use 61 Free 155 Disabled 0
Calls : Modem 5 Digital 0 Voice 0 Fax-relay 0

the box has two NP108 and one 8E1, but if it uses 61 DSP lines on 11
calls we won't be remotely able to fill the lines. Does active just mean
"powered because it was used at some time" or is it used to an incoming
call can't use it anymore?

We run IOS 12.4(1c) as preinstalled on the boxes.

Regards,
Bernhard

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RE: AS5350XM in mixed VoIP and dialup environment [ In reply to ]
Bernhard,

There are many questions there, and some, such
as the number of DSP in use, may be better handled
by a call to TAC.

First, you need to have a separate Voice DSP license
before you can proceed with accepting VoIP calls on
the AS5350.

Second, the method that you would use for VoIP calls
would be based on dialpeers. The dialpeer matching
would occur BEFORE the Group-Async command was used.

The configuration for a VoIP GW is VERY different then
a standard NAS box.

You would generally use dialpeers and not dialer interfaces.
Dialer does not apply at all to VoIP calls.

Calls can be TDM hairpinned from the E1 connect to the
PSTN to the E1 connected to the channel box and not
use DSP ONLY IF They are ISDN signalled Voice calls.
(ie CAS calls that have inband signals require a DSP.)

This architecture is well deployed and you should not
have any major problems, but you many need to upgrade
if you reach capacity issues.

Regards,
Darryl Sladden
Product Manager AS5000
Cisco Systems - ABU
dsladden@cisco.com
408-525-8970



> -----Original Message-----
> From: cisco-nas-bounces@puck.nether.net
> [mailto:cisco-nas-bounces@puck.nether.net] On Behalf Of
> Bernhard Schmidt
> Sent: Tuesday, January 03, 2006 10:51 AM
> To: cisco-nas@puck.nether.net
> Subject: [cisco-nas] AS5350XM in mixed VoIP and dialup environment
>
> Hi everyone,
>
> I sent some of those questions to cisco-nsp already, but
> maybe this (or at least for some parts cisco-voip) is the
> better forum.
>
> we're currently evaluating two Cisco AS5350XM for use in our
> university network. It should replace the old Ascend TNT
> boxes for ISDN/modem dialup and provide a SIP/PSTN gateway to
> use for our VoIP PBX to be installed next year. The whole
> setup will basically look like this
>
> +----------------+
> | PSTN |
> +----------------+
> | | | | | |
> | | | | | | 6*E1
> | | | | | |
> +----------------+ E1 +-------------+
> | AS5350XM |-----------| channel box |
> +----------------+ +-------------+
> +
> + SIP
> +
> +----------------+
> | SIP PBX (*) |
> +----------------+
>
> Current setup is just one E1 to our HiCom.
>
> All six E1 lines will be configured the same way and will
> have a large block as well as several additional numbers
> configured (so calls to one number can be signalled on one random E1).
>
> Data (ISDN) or Voice (modem) calls to several numbers on that
> trunk should be handled by the box itself (ordinary PPP
> dialup). A small block of our numbers should be sent to the
> channel box (so that is basically PSTN-to-PSTN switching). An
> important thing here would be that PSTN to the channel box is
> transparent regarding data, so we can connect any device
> there. All remaining destinations should be sent to the PBX with SIP.
>
> My first question is regarding dialup. Currently we have the
> whole dialup configuration on Serial3/0:15 and additionally
> on Group-Async0, which has both 108port spes configured in
> (group-range 1/00 2/107). This makes the Cisco answer each
> and every call it receives with PPP. If I wanted to connect
> to different "configuration profiles" depending on the dialed
> number, I had to put a "dialer pool-member x" on the lines
> and use it in several Dialer-interfaces with the same "dialer
> pool x" and different "dialer called <number>", correct?
> Would that work with async
> (modem) connections as well? As far as I understood the
> documentation it won't, and I would need to use
> resource-pooling to assign specific numbers to specific async
> lines and then group those lines specifically.
>
> The second question is about the use of the SPE dsps. We
> patched some lesser used lines/numbers to the two Ciscos we
> got and the output of "sh spe" does not look to good. E.g. at
> the moment we have 5 modem users (no compression on the
> group-async and no multilink) and 6 digital users (multilink
> and compression allowed), of which three are ordinary
> dialups, two have multilink enabled but only one channel in
> use and one has two channels. Still, show spe shows
>
> Ports : Total 216 In-use 61 Free 155 Disabled 0
> Calls : Modem 5 Digital 0 Voice 0 Fax-relay 0
>
> the box has two NP108 and one 8E1, but if it uses 61 DSP
> lines on 11 calls we won't be remotely able to fill the
> lines. Does active just mean "powered because it was used at
> some time" or is it used to an incoming call can't use it anymore?
>
> We run IOS 12.4(1c) as preinstalled on the boxes.
>
> Regards,
> Bernhard
>
> _______________________________________________
> cisco-nas mailing list
> cisco-nas@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-nas
>

_______________________________________________
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cisco-nas@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-nas
Re: AS5350XM in mixed VoIP and dialup environment [ In reply to ]
Darryl Sladden (dsladden) wrote:

Hi Darryl,

> There are many questions there, and some, such
> as the number of DSP in use, may be better handled
> by a call to TAC.

I'll try to get something there, but since those boxes are in a "Try and
Buy" contract I don't know what support options we have at the moment.

> First, you need to have a separate Voice DSP license
> before you can proceed with accepting VoIP calls on
> the AS5350.

Understood. We got the Voice-Bundles, so I assume this is okay. Plus we
already did successful VoIP connections between E1 and SIP.

> Second, the method that you would use for VoIP calls
> would be based on dialpeers. The dialpeer matching
> would occur BEFORE the Group-Async command was used.

So the call goes through the dialpeers. If no dialpeer or a "dialpeer
data x pots" with service data_dialpeer (?) is matched the call is
handed off to the usual dialin, which would be configuration-wise either
Se3/x:15 for digital calls or Group-Async for modem calls. Or, a Dialer
interface if we had the correct configuration options on Se3/x:15 and Async.

> The configuration for a VoIP GW is VERY different then
> a standard NAS box.

Is this generally a problem? In the end we will end up with two AS5350XM
and about 14 E1 in total. Voice traffic will be less than one E1 in peak
and I don't want to dedicate one box to voice and one to data for
because of redundancy,

> You would generally use dialpeers and not dialer interfaces.
> Dialer does not apply at all to VoIP calls.

Understood, but what if I want to have several dialin numbers (that are
not handled by PSTN-VoIP, but by the modem/ISDN part) with different
profiles? At last if I want to have different authentication options per
called number (on the same E1) I'm probably stuck with Dialers. Is this
even possible at all?

> This architecture is well deployed and you should not
> have any major problems, but you many need to upgrade
> if you reach capacity issues.

I don't care about upgrading as long as reaching capacity issues is
something near either 8*E1 or 216 voice or modem calls. The majority
(maybe 80%) of our calls is digital dialin, as long as the box can
handle that without eating up tens of DSPs per connection we're fine.

Regards,
Bernhard
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