Cisco Meeting Server 2 0 H323 Gateway Deployment Guide
Cisco Meeting Server 2 0 H323 Gateway Deployment Guide
3 Call Testing 16
3.1 Inbound call from an H.323 endpoint registered to an H.323 Gateway 16
3.2 Inbound call from an unregistered H.323 endpoint by dialing an IP address 18
3.3 Inbound call from an unregistered H.323 endpoint by dialing <space_uri>@IP
address 19
3.4 Outbound call to a registered H.323 endpoint 21
3.5 Outbound call to an unregistered H323 endpoint by dialing an IP address (call
routed with H.323 Gatekeeper) 22
3.6 Outbound call to an unregistered H323 endpoint by dialing an IP address (call
routed without H.323 Gatekeeper) 25
3.7 Calling non Cisco Meeting App users from H.323 endpoint 26
4 Troubleshooting Tips 29
Figures:
Figure 1: Single combined Cisco Meeting Server deployment with H.323 Gateway . 4
1 Introduction
The Cisco Meeting Server was formerly called the Acano Server. The Cisco Meeting Server is
now hosted on a new preconfigured version of a Cisco UCS server, called the Cisco Meeting
Server 1000. It can also be hosted on the Acano X-Series hardware, or on a specification based
VM server. The Cisco Meeting Server software is referred to as the Meeting Server throughout
the remainder of this guide.
Note: coSpace has been renamed space. This document has been changed to using space,
except where it refers to coSpace API objects.
The Meeting Server includes an H.323 Gateway component. This gateway is designed to be
used only with the Call Bridge of the Meeting Server, as shown below. Once the call reaches the
Call Bridge it is processed according to the normal dial plan rules. Outgoing calls from the Call
Bridge devices can also be made to H.323 devices. If the call is into a space on the Meeting
Server then it stops at the Call Bridge.
Figure 1: Single combined Cisco Meeting Server deployment with H.323 Gateway .
This guide covers one of the recommended deployments (a single combined server with the
H.323 Gateway enabled on the same Meeting Server as the Call Bridge) – as shown above.
In a split deployment we recommend deploying the H.323 Gateway as a core component, to
ensure there is no firewall between the H.323 Gateway and the Call Bridge. Typically, the H.323
Gateway will be deployed on the same Meeting Server as the Call Bridge. However, the H.323
Gateway may be deployed on a separate Core server to the Call Bridge, which is useful for test
purposes.
In a scalable & resilient deployment you can enable one H.323 Gateway per Call Bridge, again
either on the same or separate Meeting Servers.
The commands to configure and enable the H.323 Gateway are the same in all deployments,
although the IP address for the H.323 Gateway to connect differs if the Call Bridge and the
H.323 Gateway are on the same host, compared to them being on different hosts, see section
2.2, step 6.
Note: A certificate is required on each H.323 Gateway . The certificate can be signed by an
internal CA. Follow the steps in Appendix B to assign a certificate/private key pair to the H.323
Gateway.
1.1.1 Commands
In this document, commands are shown in black and must be entered as given—replacing any
parameters in <> brackets with your appropriate values. Examples are shown in blue and must
be adapted to your deployment.
2.1 Overview
The H.323 Gateway is a core component and the recommended deployment is to deploy it on
the same server as the Call Bridge. However, the H.323 Gateway can be enabled on a separate
Core server.
The H.323 Gateway listens on a minimum of two ports:
n For H.323 incoming calls to be interworked to SIP for forwarding to the Call Bridge
n For SIP incoming calls from the Call Bridge to be interworked to H.323 before being
forwarded
We recommend that the H.323 Gateway listens on the same interface as the Call Bridge (but on
different port numbers) and is used to listen for both SIP and H.323 calls, see Figure 3
For external outgoing calls from the gateway, we recommend that the H.323 calls are
forwarded to a H.323 Gatekeeper (see Figure 4) that deals with routing e.g. using dial plan rules.
Note: the H.323 Gateway supports a single next hop. Use the H.323 Gatekeeper’s neighboring
feature to allow the H.323 Gateway to reach multiple gatekeepers.
In order to accept calls from H.323 endpoints and make calls to them, the Call Bridge must be
configured to use the H.323 Gateway as the destination for outgoing calls via the Outbound dial
plan, and the H.323 Gateway must be configured to forward SIP calls to a Call Bridge.
This section provides example configurations for the recommended deployment that must be
adapted to your topology.
Note: Some old H.323 gatekeepers do not provide the domain in destination addresses for
outgoing calls, and cannot handle a domain for incoming calls. These gatekeepers supply and
require an E.164 address or H.323 id for the address.The commands to support these devices,
are listed in Appendix C.
2.1.1 Prerequisites
The instructions in this section assume that the other components of the Meeting Server have
been set and are running.
n The ports in Table 1 are required by the H.323 Gateway (see Figure 5).
Traffic
direction with Link
Connecting Destination respect to shown in Additional
Component to port to open Traffic type component Figure 5 information
Follow these steps to configure and enable the H.323 Gateway component on the appropriate
Meeting Server. If your deployment does not use an H.323 Gatekeeper, omit step 5.
1. SSH into the MMP and log in.
2. Configure the interfaces that the H.323 Gateway listens on for incoming H.323 calls (call
flow number 2 in Figure 5).
The command h323_gateway h323_interfaces <interface whitelist> allows
you to configure the interfaces that the H.323 Gateway listens for H.323 traffic on (chosen
from A, B, C or D). By default the H.323 Gateway listens on no interfaces. For example,
configure the h323_interfaces to listen on interface A, enter:
h323_gateway h323_interfaces a
3. Configure the interfaces that the gateway listens on for incoming SIP calls from the Call
Bridge (call flow number 6 in Figure 5).
The command h323_gateway sip_interfaces <interface whitelist> allows
you to configure the listening interfaces (for 1.7.0 this is interface A). By default the sip_
interfaces listens on no interfaces.
For example, configure the sip interface to listen on interface A
h323_gateway sip_interfaces a
4. Configure the ports for the SIP interface to listen on (call flow number 6 in Figure 5). By
default the H.323 Gateway uses 6061.
Note: if you wish to change the default port from 6061, and if the H.323 Gateway and Call
Bridge are on the same server, make sure you avoid port 5061 which is used by the Call
Bridge.
5. Configure the H.323 Gatekeeper’s (call control’s) hostname or IP address (call flow number
3 in Figure 5).
The H.323 Gateway will connect to this IP address for all outgoing H.323 calls and let the call
control device handle the routing.
For example, if the gatekeeper in the figure above is at IP address 192.168.1.110
h323_gateway h323_nexthop 192.168.1.110
Note: omit this step, if your deployment does not use an H.323 Gatekeeper.
6. Configure the Call Bridge IP address (call flow number 5 in Figure 5).
7. The H.323 Gateway will connect to this IP address for all outgoing SIP calls and let the Call
Bridge handle the routing via its dial plan. From R1.8, if the Call Bridge and the H.323
Gateway are on the same host then you must use IP address 127.0.0.1.
l If Call Bridge and H.323 Gateway on the same host, use:
h323_gateway sip_proxy 127.0.0.1
l If the Call Bridge and H.323 Gateway are on the different hosts then set the IP address
to be the address of the Call Bridge, which must be reachable from the H.323
Gateway.
Example of Call Bridge and H.323 Gateway on the different hosts, the IP address of
the Call Bridge being 192.168.6.25
h323_gateway sip_proxy 192.168.6.25
8. Enable the H.323 Gateway component.
h323_gateway enable
Use the command h323_gateway to check the configuration. A typical output is shown in
Figure 6.
A certificate is required on each H.323 Gateway; the certificate can be signed by an internal CA.
Follow the steps in Appendix B to generate the private key and signed certificate, then upload to
each H.323 Gateway.
Note: if the H.323 Gateway and Call Bridge are on the same host then use IP address
127.0.0.1:6061. If they are on different hosts, then use one of the external interfaces that
the Call Bridge is listening on.
Note: The Meeting Server's Outbound dial plan rule defaults to using port 5061, if not
specified. You need to change this to match the port used by the SIP interface of the H.323
Gateway, which defaults to 6061.
Encryption: Encrypted
Figure 7: Web Admin Interface showing Outbound dial plan with match all domains
To divert calls with a specific destination domain through the H.323 Gateway, create an
Outbound dial plan rule for the Call Bridge as follows:
1. Sign in to the Meeting Server’s Web Admin Interface .
2. Select Configuration>Outbound calls.
3. Complete the following fields:
l Domain: The destination domain. In the example below, the domain is @h323.com. All
calls to <anything>@h323.com will be diverted through the H.323 Gateway.
l SIP Proxy to Use: Specify the SIP interface of the H.323 Gateway. If the H.323 Gateway
and Call Bridge are on the same host then use IP address 127.0.0.1:6061. If they are on
different hosts, then use one of the external interfaces that the Call Bridge is listening on.
Note: The Outbound dial plan rule defaults to using port 5061, if not specified. You need
to change this to match the port used by the SIP interface of the H.323 Gateway, which
defaults to 6061.
l Encryption: Encrypted
Figure 8: Web Admin Interface showing Outbound dial plan with match domain @h323.com
Note: the H.323 Gateway cannot modify the dialed address. If you require dialed addresses to
be modified, then you will need to use an H.323 Gatekeeper in your deployment that is capable
of modifying dialed addresses.
user@domain – which is processed by the Call Bridge Incoming dial plan rules; can be a
space, a Cisco Meeting App user or even an external user.
Note: set call matching for <IP address> or the <domain> on the Web Admin Interface.
3 Call Testing
Depending on the type of call(s) or meetings you intend to have, read and follow the example(s)
in the appropriate section(s).
n Inbound call from an H.323 endpoint registered to an H.323 Gateway
n Inbound call from an unregistered H.323 endpoint by dialing an IP address
n Inbound call from an unregistered H.323 endpoint by dialing <coSpace_uri>@IP address
n Outbound call to a registered H.323 endpoint
n Outbound call to an unregistered H323 endpoint by dialing an IP address (call routed with
H.323 Gatekeeper)
n Outbound call to an unregistered H323 endpoint by dialing an IP address (call routed without
H.323 Gatekeeper)
n Calling non Cisco Meeting App users from an H.323 endpoint
Note:
1) H.323 signaling is unencrypted, so you can use pcap to obtain an H.323 trace. In addition, the
MMP command h323_gateway trace_level <level> provides additional logging to aid
troubleshooting by Cisco support. You may be asked to provide traces for levels 0, 1 or 2.
2) Support for legacy gatekeepers and endpoints: some old H.323 gatekeepers do not provide
the domain in destination addresses for outgoing calls, and cannot handle a domain for
incoming calls. These gatekeepers supply and require an E.164 address or H.323 id for the
address. There are commands to support these devices, seeAppendix C"H.323 Gateway
address handling for older gatekeepers " on page 34 for more details.
Figure 9: Call Flow for Inbound Call from Registered H.323 Endpoint
Where:
Note: From the Call Bridge's point of view, this is an incoming SIP call from the H.323 Gateway .
After the call has connected, check Status>Calls in the Web Admin Interface (see Figure 10)
The call status on the H.323 Gatekeeper, which in this example is a Cisco VCS, shows it as a
H.323 call (see Figure 11). The Meeting Server's H.323 Gateway performs the interworking.
Note: By changing the default URI to an IVR URI, after the incoming call is connected, users
can enter a space’s Call ID to join a specific space.
n For IP dialing to work for calls from both internal and external H.323 endpoints, use the H.323
Gateway as a Core component but add a second listening interface for the external calls e.g.
on interface B.
h323_gateway h323_interfaces a b
Note: the H.323 Gateway cannot traverse NAT or firewalls. The second listening interface for
external calls must be internet facing with a public IP address.
Example setup:
n Calling party’s unregistered H.323 endpoint name: Brian.ex60
n Dialing: 192.168.1.91
Figure 12: Call flow for inbound call from unregistered H.323 endpoint
Where:
After the call has connected, check the call status in the Web Admin Interface (see Figure 13).
n For IP dialing to work for calls from both internal and external H.323 endpoints, use the
H.323 Gateway as a Core component but add a second listening interface for the external
calls e.g. on interface B
h323_gateway h323_interfaces a b
Note: the H.323 Gateway cannot traverse NAT or firewalls. The second listening interface
for external calls must be internet facing with a public IP address.
Example setup:
n Calling party’s unregistered H.323 endpoint name: example.MXP
n Dialing: [email protected]
Example setup:
n Calling party’s Cisco Meeting App user URI: [email protected]
n Dialing: [email protected]
Figure 16: Call flow for outbound call to a registered H.323 endpoint
Where:
After the call has connected check the status in the Web Admin Interface (see Figure 17).
The call status on the H.323 Gatekeeper shows it as an H.323 call, in this example the H.323
Gatekeeper is a Cisco VCS, (see Figure 18). The Meeting Server's H.323 Gateway performs the
interworking.
Note: in this scenario, the H.323 endpoint is unregistered, but an H.323 Gatekeeper routes the
call.
Prerequisite:
n Decide on which dial plan rule to use, default rule or a custom rule
n On the H.323 Gatekeeper (in this example this is a Cisco VCS), in the Dial plan
configuration set the ‘Calls to Unknown IP Addresses’ to Direct (see Figure 19) and ensure
that the H.323 Gatekeeper can reach the endpoint that you are dialing.
n For IP dialing to work for calls from both internal and external H.323 endpoints, use the
H.323 Gateway as a Core component but add a second listening interface for the external
calls e.g. on interface B
h323_gateway h323_interfaces a b
Example setup:
n Calling party’s Cisco Meeting App user URI: [email protected]
n Dialing: 192.168.1.219
Figure 20: Call flow for outbound call to an unregistered H.323 endpoint by dialing an IP address
Where:
When the call has connected, check the status in the Web Admin Interface (see Figure 21)
If you are using a Cisco VCS as the H.323 Gatekeeper, the call will look similar to Figure 22.
Note: in this scenario, the H.323 endpoint is unregistered, and there is no H.323 Gatekeeper to
route the calls.
Prerequisite:
n The h323_gateway h323_nexthop configuration must be removed if previously set. For
example:
a. SSH into the MMP and log in.
b. Remove the h323_gateway h323_nexthop configuration
h323_gateway del h323_nexthop
n For IP dialing to work for calls from both internal and external H.323 endpoints, use the
H.323 Gateway as a Core component but add a second listening interface for the external
calls e.g. on interface B
h323_gateway h323_interfaces a b
Note: the H.323 Gateway cannot traverse NAT or firewalls. The second listening interface
for external calls must be internet facing with a public IP address.
Example setup:
n Calling party’s Cisco Meeting App user URI: [email protected]
n Dialing: 192.168.1.201
Figure 23: Call Flow for Outbound Call to a Unregistered H.323 Endpoint
Where:
When the call has connected, check the status in the Web Admin Interface (see Figure 24).
3.7 Calling non Cisco Meeting App users from H.323 endpoint
It is possible to configure the Call Bridge dial plan to be able to call Lync or SIP users. These calls
are transcoded by the Call Bridge.
In this example both the H.323 Gateway and Call Bridge act as a gateway. It is assumed that the
Call Bridge dial plan already allows forwarding of SIP calls to Lync. The H.323 call is interworked
into a SIP call to the Call Bridge which then forwards the call to Lync. In this case the Call Bridge
handles all transcoding of media.
Example setup:
n Calling party’s registered H323 endpoint alias: [email protected]
n Dialing: [email protected]
where lyncuser1.lync@exampledemo is the Lync address for the called
Figure 25: Call flow from H.323 endpoint to a non Cisco Meeting App user
Where:
When the call has connected check the status in the Web Admin Interface (see Figure 26).
If you are using a Cisco VCS as the H.323 Gatekeeper, the call will look similar to Figure 27.
4 Troubleshooting Tips
n After changing the configuration of the H.323 Gateway , the gateway may not work as
expected. Reboot the Meeting Server to solve the issue.
n The Meeting Server's Outbound dial plan rule defaults to using port 5061, if not specified.
You need to change this to match the port used by the SIP interface of the H.323 Gateway ,
which defaults to 6061.
n There is a bug in the Cisco VCS software. Sometimes the settings in Zone > Custom aren't
saved. You may need to delete the zone and start again - remember to update the dial plan
search rules to match.
n The MMP command h323_gateway trace_level <level> provides additional logging
to aid troubleshooting by Cisco support. You may be asked to provide traces for levels 0, 1 or
2.
Note: the public key is created and held within the .csr file.
2. Submit the .csr file to the CA (public CA or internal CA) for signing
You can use the pki command on the Meeting Server to generate the private key and .csr file,
and submit the pair to an internal CA such as an Active Directory server with the Active
Directory Certificate Services Role installed.
3. SSH into the MMP
4. Upload the signed certificate and intermediate CA bundle (if any) to the Meeting Server
using SFTP.
5. Check that the certificate (and certificate bundle) and the private key match
pki verify <certicate> <cert bundle/CA cert> [<CA cert>]
6. Assign the certificate (and certificate bundle) and private key pair to the H.323 Gateway .
h323_gateway certs <keyfile> <certificatefile> [<cert-bundle>]
7. Restart the H.323 Gateway
h323_gateway restart
Figure 29 illustrates how call back can be made to work for calls originating from legacy H.323
gatekeepers. Using command h323_gateway h323_domain_strip yes removes the
domain from the destination if it matches h323_domain and using command h323_gateway
sip_domain_strip yes removes the domain from the source address if it matches sip_
domain, when making a call to the legacy gatekeeper.
Note: dialing an IP address does not require removal of the domain, as the Call Bridge
recognizes that it is an IP address and does not append the domain.
Command/Examples Description/Notes
h323_gateway certs Defines the name of the private key file and .crt file for the
<keyfile> <certificate
H.323 Gateway application and, optionally, a CA certificate bundle as provided by
file> [<cert-bundle>]
your CA. (See "Assigning a certificate/private key pair to the H.323 Gateway" on
page 33.)
h323_gateway h323_ Connect to this IP address for all outgoing H.323 calls and let the device at this IP
nexthop <host/ip> address handle the routing. If this address is not set, only IP dialing works.
h323_gateway del h323_
nexthop Typically this IP address is a Cisco VCS/Polycom DMA, and an H.323 trunk is
established between the Meeting Server's H.323 Gateway and the third party
device (H.323 Gatekeeper).
The H.323 Gateway does not register with the device, just forwards calls to them
– the device will need to be configured appropriately to accept these calls.
h323_gateway default_ Optional. If an incoming H.323 call has no destination (normally only the case
uri <uri> when the H.323 Gateway has been dialed by an IP address) the SIP call is made
to whatever default_uri is set. The default_uri may point to an IVR, or directly into
a space. If it is not set, the call is rejected.
h323_gateway del
default_uri
h323_gateway sip_ Optional. If an incoming H.323 call is made to the gateway without a domain in
domain <sip_domain > the destination address, @<sip_domain> will be appended to the destination
address before the SIP call to the Call Bridge is made.
h323_gateway sip_ If set to "yes" and "h323_gateway sip_domain" is set, when a SIP call is made to
domain_strip <yes/no> the gateway the @<sip_domain> will be stripped from the source address (if
present) before making the H.323 call.
h323_gateway h323_ Optional. If an H.323 call is made to the gateway without including a domain in
domain <h323_domain> the source address, @<h323_domain> will be appended to the source address
h323_gateway del h323_ before the SIP call is made.
domain
Command/Examples Description/Notes
h323_gateway h323_ If set to "yes" and "h323_gateway h323_domain" is set, when a SIP call is made
domain_strip <yes/no> to the gateway the <h323_domain> will be stripped from the destination address
(if present) before making the H.323 call.
h323_gateway h323_ Must be configured in order for the gateway to start, but the actual setting is
interfaces <interface currently ignored.
list>
h323_gateway sip_
interfaces <interface
list>
h323_gateway sip_port Ports for the SIP side to listen on. The default is 6061.
<port>
Note: if you wish to change the default port from 6061, and if the H.323 Gateway
and Call Bridge are on the same server, make sure you avoid port 5061 which is
used by the Call Bridge. Changes do not take place until the gateway is restarted.
h323_gateway sip_proxy Set this to the IP address of the Call Bridge, or for multiple Call Bridges use the
<uri> domain name (through DNS). All incoming H.323 calls will be directed to this uri
h323_gateway restrict_ If set to yes, the H.323 Gateway is limited to a safe set of codecs that are less
codecs <yes/no> likely to cause interoperability problems. Currently this set is
G.711/G.722/G.728/H.261/H.263/ H.263+/H.264.
Codecs disabled by this feature are G.722.1 and AAC.
h323_gateway trace_ Provides additional logging to aid troubleshooting by Cisco support. You may be
level <level> asked to provide traces for levels 0, 1, 2 or 3.
PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE
FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.
THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT
ARE SET FORTH IN THE INFORMATION PACKET THAT SHIPPED WITH THE PRODUCT AND ARE
INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE
SOFTWARE LICENSE OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE
FOR A COPY.
The following information is for FCC compliance of Class A devices: This equipment has been
tested and found to comply with the limits for a Class A digital device, pursuant to part 15 of the
FCC rules. These limits are designed to provide reasonable protection against harmful
interference when the equipment is operated in a commercial environment. This equipment
generates, uses, and can radiate radio-frequency energy and, if not installed and used in
accordance with the instruction manual, may cause harmful interference to radio
communications. Operation of this equipment in a residential area is likely to cause harmful
interference, in which case users will be required to correct the interference at their own
expense.
The following information is for FCC compliance of Class B devices: This equipment has been
tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the
FCC rules. These limits are designed to provide reasonable protection against harmful
interference in a residential installation. This equipment generates, uses and can radiate radio
frequency energy and, if not installed and used in accordance with the instructions, may cause
harmful interference to radio communications. However, there is no guarantee that interference
will not occur in a particular installation. If the equipment causes interference to radio or
television reception, which can be determined by turning the equipment off and on, users are
encouraged to try to correct the interference by using one or more of the following measures:
l Reorient or relocate the receiving antenna.
l Increase the separation between the equipment and receiver.
l Connect the equipment into an outlet on a circuit different from that to which the receiver
is connected.
l Consult the dealer or an experienced radio/TV technician for help.
Modifications to this product not authorized by Cisco could void the FCC approval and negate
your authority to operate the product.
The Cisco implementation of TCP header compression is an adaptation of a program
developed by the University of California, Berkeley (UCB) as part of UCB’s public domain version
of the UNIX operating system. All rights reserved.
Copyright © 1981, Regents of the University of California.
NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND
SOFTWARE OF THESE SUPPLIERS ARE PROVIDED “AS IS” WITH ALL FAULTS. CISCO AND THE
ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED,