About H.323 gatekeepers and SIP proxies
You can configure the Pexip Infinity platform with the addresses of one or more H.323 gatekeepers and SIP proxies. These are the call control systems that can be used to route outbound H.323 and SIP calls on behalf of Pexip Infinity.
The H.323 gatekeepers and SIP proxies configured here are used for outbound calls from Pexip Infinity. They do not determine the systems used to route inbound calls to Pexip Infinity. Pexip Infinity will not automatically register with any systems configured here. To route inbound calls from a gatekeeper or SIP proxy, you must configure these systems with neighbor zones that direct calls to Pexip Infinity. For more information see Call control.
Outbound calls are made from Pexip Infinity when:
- a conference participant uses a Connect app to add another participant to the call
- the administrator manually dials out to a participant from a Virtual Meeting Room
- a participant is automatically dialed out to from a Virtual Meeting Room
- a third party uses the API to place a call to a participant
- the Infinity Gateway is used to interwork a call.
Nominating which H.323 gatekeeper and SIP proxy to use
To configure H.323 gatekeepers, go to , and to configure SIP proxies, go to .
When configuring the proxy or gatekeeper, the target Address can be an IP address or an FQDN, and for a SIP proxy you must also specify the transport protocol.
If the target address is an FQDN, Pexip Infinity looks for DNS SRV records first if the Port field is blank. If a port is specified then Pexip Infinity only performs a DNS A / AAAA lookup. When using SRV records, if multiple records are returned they are used in order based on priority and weight, and if the connection to the target host(s) fails then the host(s) associated with the target of the next SRV record are tried (as per RFC3263 for SIP and H.323 Annex O).
Example DNS SRV records for locating H.323 gatekeepers and SIP proxies
This example shows the DNS SRV records required for H.323 gatekeepers and SIP proxies, which in this case are two Cisco VCSs in the vcs.example.com domain. In this example there are two h323ls service records for the two target VCS gatekeepers, and there is a set of two service records for the two target VCS SIP proxies for each transport protocol (TCP, TLS i.e. sips and UDP). Note that if the SIP proxy or gatekeeper is hosted in a remote environment, you (as the administrator for your own domain) will not have any authority over the DNS records for that remote domain — and are not responsible for creating them — you should check that such records exist when troubleshooting any outbound calling issues.
| Name | Service | Protocol | Priority | Weight | Port | Target host |
|---|---|---|---|---|---|---|
| vcs.example.com. vcs.example.com. |
h323ls h323ls |
udp udp |
10 10 |
10 10 |
1719 1719 |
vcs01.vcs.example.com. vcs02.vcs.example.com. |
| vcs.example.com. vcs.example.com. |
sip sip |
tcp tcp |
10 10 |
10 10 |
5060 5060 |
vcs01.vcs.example.com. vcs02.vcs.example.com. |
| vcs.example.com. vcs.example.com. |
sips sips |
tcp tcp |
10 10 |
10 10 |
5061 5061 |
vcs01.vcs.example.com. vcs02.vcs.example.com. |
| vcs.example.com. vcs.example.com. |
sip sip |
udp udp |
10 10 |
10 10 |
5060 5060 |
vcs01.vcs.example.com. vcs02.vcs.example.com. |
In your actual deployment, both the Name and the Target host should be changed to use the real domain name and host names of the call control systems.
After adding the details of the H.323 gatekeeper or SIP proxy, you must then associate it with the relevant location () or Call Routing Rule ().
- Each System location can have a nominated H.323 gatekeeper and SIP proxy — these are used when adding a new H.323 or SIP participant to a conference and define where to route the outbound H.323/SIP calls placed from nodes within that location.
- Each Call Routing Rule can have a nominated H.323 gatekeeper or SIP proxy — these are used to route the outbound leg of gateway calls for rules that are targeted at external H.323 and SIP systems.
If a Connect app user adds an H.323 or SIP participant to a conference, the call is routed to the H.323 gatekeeper or SIP proxy that is associated with the system location of the Conferencing Node to which the Connect app is connected. If the location does not have a configured H.323 gatekeeper or SIP proxy, Connect app users connected to Conferencing Nodes in that location may still be able to dial in other participants to a conference — to enable this, the user must enter the participant's IP address or FQDN (and for the latter, DNS must be configured properly), thus allowing Pexip Infinity to dial the participant directly.