VoIP Trunking On-Net Call
This call type occurs between endpoints connected to Cisco Unified Communications
Manager and a Legacy PBX that is connected to a Voice Gateway. The call type includes:
- SIP/SCCP signaling traffic between the endpoint and the Cisco Unified
Communications Manager
- SIP signaling traffic between the Voice Gateway and the Cisco Unified
Communications Manager
- TDM signaling traffic between the Legacy PBX and Voice Gateway
- Media traffic between the endpoint and Voice Gateway
On-Net Call (VoIP Trunking)
Usage |
- Called number must follow the enterprise
internal numbering plan requirement
- Called number must be defined by ranges and
not individual numbers
- Called number can be either DN or ISP +DN
depending on the enterprise internal
numbering plan adopted. DN can be either just
Extension for flat Dial Plan or SLC +
Extension or ISP + SLC + Extension.
- Voice codec used must be selected per site
- Only voice calls can be made; video calls are
not supported
- Fax is supported as best effort only
- MoH is provided in accordance with the site’s
MoH policy
- Voice Gateway configuration is part of the
solution
- Voice Gateway redundant deployment is not
supported
- Enbloc signaling is between the Voice Gateway
and Cisco Unified Communication Manager
- Any target number can be used unless
restricted by Class of Service
- Codec is dynamically selected based on the
endpoints used
- Alternate call routing when the Legacy PBX or
Voice Gateway is unreachable is not supported
|
Accessibility |
- User can perform On-Net call from any
endpoint registered with Cisco Unified
Communications Manager
- Legacy PBX connected using a Voice Gateway is
considered to be similar to an Inter-Site
call
- User uses the same dialing behavior as
Inter-Site On-Net Call
|
Usage Example |
Enables the integration with the existing
environment during the transition period of all
users |
Default
Configuration |
- Available to all users at all sites of the
enterprise
- Codec: Voice - G.729 and G.711
- Codec: Sample Size - 20ms/20Bytes and
20ms/160Bytes
- Bandwidth: 8kbps and 64kbps
|
Configuration
Choices |
- Feature availability cannot be changed by
site or user
- Codec can be selected between G.711 and G.722
on a per-site basis
|
Redundancy |
Available to users without restrictions |
Survivability |
Not available to users in fallback mode |
Endpoint Types
Supported |
- Cisco IP phones
- Cisco ATA
- Cisco VG
|
Example |
Example:
Phone D (DN=8 200 6100)
Dial 8 400 1234 to Legacy PBX
Connected Number shows 8 400 1234
Legacy PBX (Site 3)
Dial 8 100 2123 to Call A
Phone C (DN=8 100 2123)
|