[cisco-voip] MGCP vs H323
Wes Sisk
wsisk at cisco.com
Wed Aug 30 22:13:44 EDT 2006
You will get calling name with h.323 4.2 depending on how it is
delivered:
You will see it if delivered by:
DisplayIE
NotifyIE
You will not see it if it is delivered by FacilityIE. Delivery by
FacilityIE is the most common form we encounter today. This is
commonly referred to as 'c-nam', 'cname', or 'ni-3 name delivery'.
/Wes
On Aug 30, 2006, at 3:21 PM, Linsemier, Matthew wrote:
This has been an issue since we installed CallManager almost 4 years
ago. If Cisco is going to move people in a new direction H.323/SIP
it certainly needs to support the same feature set. Can anyone
confirm that they can get Calling Name in CCM 4.2? We have had
calling name in MGCP for all this time and to remove this would cause
serious havoc amongst my users.
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
Sent: Wednesday, August 30, 2006 1:55 PM
To: Jason Burwell; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP vs H323
H323 does support calling party number, just not name. Like Jason
mentioned before, the options are there and you can see name traverse
inbound in debugs, but the gateway does not send it on to CallManager.
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Jason Burwell
Sent: Wednesday, August 30, 2006 1:36 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP vs H323
We have just purchased CVP and are being told that we can no longer
use MGCP gateways and must migrate to H323. According to what I have
heard here, H323 will not support Caller Name but will H323 support
Calling Party Number capability?
Jason
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Jason Aarons (US)
Sent: Wednesday, August 30, 2006 12:52 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP vs H323
I believe the issue is with CallManager not supporting both types of
facility ie as you can see the Calling Party Name in the router’s
debug isdn q931.
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Linsemier, Matthew
Sent: Wednesday, August 30, 2006 12:11 PM
To: Matt Slaga (US); Joe Pollere (US); Nick Kassel; cisco-
voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP vs H323
Calling Name/Number is one of the main reasons we utilize MGCP over H.
323. This coupled with failover (which seems to be handled much more
gracefully then in an H.323 environment) keep us on this protocol.
If H.323 could handle Calling Name, I would definitely make the
move. It would certainly make fax integration with products such as
Unity or XMedius fax much easier. I, however, find myself in a
position where I couldn’t possibly take away features that my users
have come to expect and love.
I did hear rumblings that H.323 was going to support Calling Name in
a later IOS release. The information is there on the router, just
not getting processed. Can any of you Cisco folks comment (off the
record)?
-Matt
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
Sent: Wednesday, August 30, 2006 12:00 PM
To: Joe Pollere (US); Nick Kassel; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP vs H323
At this point, MGCP is able to provide name/number display on a PRI
(if provided from the carrier) whereas this is not yet possible in H323.
I have started to hear some rumblings within Cisco about MGCP being
slowly phased out and eventually being replaced with SIP and/or
H323. Sounds about right since Cisco has been shoving MGCP down
everyone’s throat for the past three years that they would want to
keep this down to a rumbling.
Personally, I would only use MGCP when interconnecting with a PBX
during a migration to keep from having thousands of dial peers. Or,
if I needed name/number display on a PRI.
Funny thing is (and this may be fixed now), but with FXO-M1s the only
way to get name/number display was with H323. MGCP couldn’t do it.
Sounds kind of backwards that it’s just the opposite with a PRI
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Joe Pollere (US)
Sent: Wednesday, August 30, 2006 10:51 AM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP vs H323
Nick,
This is from the GWGK course book:
Using H.323 as the call control protocol to a gateway has the
following advantages:
H.323 provides integrated access. Data and voice channels can be
placed on the same T1.
For example, for a service provider like AT&T, FR and PRI can be
placed on the same T1.
H.323 provides support for fractional PRI.
Gateways support caller ID on FXO ports. CallManager does not support
caller ID on FXO
ports from MGCP gateways.
Many more TDM interface types and signaling protocols—for example,
analog-Direct
Inward Dialed (DID), recEive and transMit (E&M), T1 Feature Group-D
(FGD), and E1
R2—can be used.
H.323 drops DSPs on hairpinned calls to enable capabilities like ISDN
video switching.
Gateway resident applications like Toolkit Command Language (TCL) and
voice
extensible markup language (VXML) can be used. TCL and VXML
applications provide
IVR features and call control functionality such as call forwarding,
conference calling, and
voice mail.
CAC network design with H.323 gatekeepers is often necessary when
voice and video
coexist in a network and Cisco CallManager is not the only call
controller in the network.
There are no release dependencies between gateways and Cisco
CallManager for
supporting new voice hardware. New hardware cards on Cisco IOS
gateways become
immediately available for use with all existing Cisco CallManager
releases.
H.323 enables a much easier migration architecture to SIP because the
fundamental
concepts of H.323 and SIP—for example, distributed control with dial-
peer
configurations—are the same.
Calls from IP phones through an H323 gateway are dropped on a
CallManager failover
unless SRST mode is enabled. With SRST enabled, the calls are preserved.
Using MGCP as the call control protocol to a gateway has the
following advantages:
Centralized configuration, control, and download from Cisco CallManager
Better feature interaction with capabilities like caller ID and name
display
Easy, centralized dial-plan management
Gateway voice security features (voice encryption) as of Cisco IOS
Software Release
12.3.(5th)T
Q Signaling (QSIG) supplementary services as supported by Cisco
CallManager:
— Cisco CallManager interconnects to a QSIG network using an MGCP
gateway and
T1 or E1 PRI connections to a private integrated services network
(PISN). The
MGCP gateway establishes the call connections. Using the PRI backhaul
mechanism, the gateway passes the QSIG messages to the Cisco
CallManager to set
up QSIG calls and send QSIG messages to control features.
— When a PBX is connected to a gateway that is using QSIG via H.323,
calls that are
made between phones on the PBX and IP phones attached to the Cisco
CallManager
can have only basic PRI functionality. The gateway that terminates
the QSIG
protocol provides only the calling line ID (CLID) and DID number,
instead of Cisco
CallManager providing that information.
Enhanced call survivability:
— Calls from IP phones through an MGCP gateway are preserved on a
CallManager
failover. This feature avoids dropped calls when applying the monthly
operating
system service release on the Cisco CallManagers
— In SRST mode, calls from IP phones through an MGCP gateway are
preserved on
MGCP fallback for calls on analog or CAS circuits. Calls on ISDN
circuits are
dropped on fallback.
HTH’s
Joe
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Wednesday, August 30, 2006 10:35 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MGCP vs H323
Does anyone know if there is a document that lists the differences or
benefits of using either MGCP or H323.
I need to install a gateway for PSTN breakout in a DR site so it
won’t be needed for redundancy or anything like that.
All our branch offices currently use H323 and we have H323 for our HQ
so that they can be used for redundancy should be an issue with the
local branch PSTN.
Just wondering whether it might be better to go with MGCP in this
instance.
Is it easier to mask the calling numbers with MGCP?
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