NEC 2000 IPS manual Network Consideration, Network Bandwidth

Chapter 13 Voice Over IP (VoIP)

Network Consideration

There are several network considerations that must be met before deployment of IP Telephony. These items are:

The network must support the Internet Protocol and the local interface must meet 802.3 Ethernet standards.

It is recommended that the managed network be based on Switched, Fast Ethernet environment for IP telephony.

Packet Delay or Latency, one-way should be less than 100 milliseconds optimal, with a maximum of 250 milliseconds one-way.

Jitter, or out of sequence packets, should be between 10 -150 milliseconds optimal with a maximum of 300 milliseconds while packet loss of 1% or less is recommended with the maximum of 3% from end to end.

Network Bandwidth

Network bandwidth requirements are based on the total number of IP trunks or IP stations installed in the PBX. The multiple algorithms available the bandwidth needed for a voice call out to an IP network can range from 10.7 Kbps to 96 Kbps.

The most common algorithm used for NEAX IP trunks is G.729A, which allows the voice to be compressed to 8KBps. Once the layer 3 (IP) overhead is added to the voice payload, the approximate bandwidth is 16KBps for a single voice stream out of the PBX to an IP network. This bandwidth calculation does not include layer 2 overhead and will vary depending on the type of transport (Frame Relay, ATM, Ethernet etc.). Depending on what type of algorithm is in place and how many devices are deployed dictates the amount of bandwidth needed.

The bandwidth for call setup for the CCIS signaling channel over IP is 7.2 Kbps per call and the FCCS signaling channel is 83.2Kbps per call. Call teardown requires the same amount of bandwidth. This number varies and 7.2 Kbps and 83.2 Kbps are averaged values from the start sequence until setup or teardown is accomplished.

This means to make an FCCS or CCIS call from PBX A to PBX B, PBX A will transmit 83.2 Kbps (FCCS) or 7.2 Kbps (CCIS) as the F/C channel until the call is established. Once the call is established, PBX A will stop sending the 83.2 Kbps/7.2 Kbps setup information and will begin sending the appropriate voice payload per the encoding/decoding algorithm selected (G.711, G.729A, G.723.1).

To make multiple calls, the system will send 7.2 Kbps/83.2 Kbps for each call until all the calls are established. During tear down 7.2 Kbps/83.2 Kbps will be transmitted for each call again.

If an IP station has been idle for up to 4 seconds, a 7.6Kbps “Keep Alive” packet will be sent to those respective devices. A “Keep Alive” packet is not generated to an idle station if it is configured across a WAN. This same process occurs on IP trunk routes. A “Keep Alive” packet is sent to the trunk route for each system or Point Code in the network. If traffic is present on either IP stations or trunk routes, no “Keep Alive” packet is generated.

UNIVERGE NEAX 2000 IPS Request for Proposal (RFP) Reference Guide	Page 13- 31
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