The Total Access 900/900e (TA 900/900e) Book-Ending Application allows you to trunk traditional TDM-based PBX phone systems together over an IP network without the need for upgrading the existing PBX systems.
In this application, a Total Access 900 or 900e unit is placed at each location, emulating either T1/PRI, or T1/E&M, or analog loop-start signaling to the customer’s PBX/Key System. The TA 900 then converts the TDM voice signaling to SIP and voice is transported via RTP over the IP Network. At the distant end, the voice is converted back to TDM for delivery to PBX or Key System.
- Eliminates the need to upgrade existing TDM-based PBX and Key Systems to support Voice over IP.
- Provides toll bypass between sites
- Provides voice connectivity over existing data circuits
- Allows trunks to host PBX to be overbooked (site to multi-site solutions)
- Preserves bandwidth for data when voice calls are not active between sites
- Disparate TDM interfaces supported, (e.g. T1/PRI at the Host PBX and FXS at the remote PBX/Key systems
- No proprietary signaling supported with PRI connections to PBX
- Site to Multi-site solution is limited to:
- Three remote sites using a TA900 at the host.
- Twelve remote sites using the TA 900e 2nd Gen at the host.
Site to Site Design:
The connection between the PBX and the Total Access 900/900e can be either T1 PRI or T1 CAS E&M signaling. T1 PRI signaling includes both Called Party (DNIS) and Calling Party (ANI) information elements in the PRI signaling. In contrast, E&M signaling only provides Called Party (DNIS) information. Therefore, PRI signaling is preferred in order to maintain caller ID (ANI) across the span.
IP connectivity between the host Total Access 900/900e and the remote Total Access 900 is required for this application. IP connectivity is configured across the WAN via Ethernet, T1, or bonded T1s (TA 9xxe only). A SIP Trunk carries all voice traffic and this configuration allows dynamic sharing of bandwidth between voice and data across the IP network.
Site to Multi-Site Design:
Using a setup similar to the site-to-site design, multiple remote sites can be connected to the host site over the IP network. The remote site TA 900 terminates the call originated from the remote site PBX and routes the call over a SIP trunk based on an internal dial plan. If T1 E&M signaling is used at the remotes, when the call arrives over the SIP trunk to the Host Site, the calling party end user would be unknown since the remote PBX does not deliver that information. However, the remote site TA 900 can substitute a name such as “Remote Site A” into the SIP packets being sent to the Host Site.
For multisite applications that exceed, or will exceed 3 remote sites, the 2nd Generation TA900e can be configured with a maximum of 12 SIP trunks. Each SIP trunk is configured for a corresponding remote site utilizing the dial plan to specify the trunk on which a particular call is to be routed. Remote site to remote site SIP signaling is always routed through the main (hub) TA 900/900e, while the RTP stream may be routed site to site based on the route table of the 9xx/9xxe or upstream router in most MPLS/PIP networks.
The network should be designed to ensure the appropriate amount of bandwidth is available at all sites, both upstream and downstream. Assume approximately 80kbps (assuming G.711 compression) or 40 kbps (assuming G.729a compression) per call multiplied times the maximum number of expected simultaneous calls over a particular link. It is advised that QoS be implemented on the networks in order to ensure voice packets take priority over other non-time sensitive data traffic. When QoS is employed, all available bandwidth could be used for voice traffic during periods of high call volume. Therefore, it is important to design the network to also be able to accommodate the minimum data bandwidth requirements during peak call times.