Planning VoIP Networks

When planning to implement Voip to a network, a network designer needs to pay particular attention to ensuring that sufficient bandwidth is available to support voice traffic on WAN links. In previous sections you already learned that the codec chosen will impact the bandwidth requirements associated with a voice call. However, other elements that need to be considered include the overhead associated with the RTP, UDP, and IP headers, as well as Layer 2 framing. In order to determine the required bandwidth, two main values first need to be calculated – the size of a voice packet, and the voice packet per second rate.


To calculate the size of a voice packet, you must add together the size of the RTP, UDP, and IP headers, as well as payload size and Layer 2 framing. For example, let’s say that you intend to use the G.729 codec without RTP header compression over a PPP link. In this case, the combined RTP/UDP/IP header size would be 40 bytes, as you learned earlier. The payload size would be 20 bytes, and the PPP framing an additional 6 bytes, adding up to 66 bytes total. Converting this number to bits yields a packet size of 66 x 8, or 528 bits total.
To determine the voice packet per second rate, divide the codec bit rate by payload size of a packet. Earlier in this chapter you learned that the G.729 codec uses a bit rate of 8 kbps, or 8000 bps. The payload of a G.729 voice packet is 20 bytes, or 160 bits. Therefore, the packet per second rate is 50 (8000 divided by 160).
Finally, to determine the bandwidth per call, multiple the total voice packet size by the number of packets per second. In this case, the calculation is 528 bits for the total packet size, multiplied by a packet per second rate of 50, for a total of 26400 bps (26.4 kbps). In other words, a call using the G.729 codec and no header compression requires approximately 26.4 kbps of bandwidth. When header compression is used (assuming a header size of 2 bytes rather than 40), the same calculation yields a bandwidth requirement of 11.2 kbps, so IP RTP header compression is definitely worth exploring. For example, if the bandwidth on the WAN link to be dedicated to VoIP traffic was 256 kbps, the link could handle approximately 9 simultaneous calls without, or 22 with IP RTP header compression. Don’t forget that voice conversations are duplex – in other words, with header compression enabled, a total of 11 simultaneous conversations across the WAN link could occur, or 11 voice data streams in each direction.


When planning a network to carry voice traffic, the numbers above provide a fairly accurate estimation of WAN bandwidth requirements. However, as you learned earlier, a typical voice call may consist of anywhere between 30 and 40 percent silence. In order to take advantage of these silences (and not transmit packets of “silence”), Voice Activity Detection (VAD) can be implemented using Cisco CallManager. When enabled, periods of silence are suppressed, and not packetized for transmission across the network. This can result in substantial bandwidth savings, which would subsequently be available for other network application traffic.

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