by these elements: Because some of these elements come from the same source, the information units that are needed for bandwidth calculation are packetization period or packetization size, codec bandwidth, IP overhead, data-link overhead, and tunneling or security overhead.
Content 2.4 Calculating Bandwidth Requirements for VoIP 2.4.2 Impact of Codecs on Bandwidth A codec transforms analog signals into digital format. Different codecs have different bandwidth requirements: The bandwidth codec, however, indicates only the bandwidth that is required for the digitized voice itself. It does not include any packetization overhead.
Content 2.4 Calculating Bandwidth Requirements for VoIP 2.4.3 How the Packetization Period Affects VoIP Packet Size and Rate The packetization overhead that a VoIP device adds to the codec bandwidth depends on the size of the added headers and the packet rate. Sending more packets adds more IP, UDP, and RTP headers to the voice payload. The overall VoIP bandwidth requirement includes the whole VoIP packet (headers and voice payload) and the sending rate of the VoIP packets. On VoIP devices, you can specify, in addition to the codec, the amount of voice encapsulated per packet. Usually, this value is configured by the packetization period (in milliseconds). A higher packetization period results in a larger IP packet size because of the larger payload (the digitized voice samples). However, a higher packetization period results in a lower packet rate, reducing the IP overhead because of the smaller number of packets that must be generated. Figure contrasts two scenarios with different packetization periods. In the first scenario, portions of 20 ms of voice (160 PCM samples) are packetized. The packet rate is the reciprocal of the packetization period. If packetization is done every 20 ms, 50 packets are generated per second. For a total of 60 ms of voice, three packets, each carrying 20 ms of voice, are needed. Therefore, packetization of this 60 ms of voice introduces an overhead of three IP, UDP, and RTP headers. In the second scenario in Figure , chunks of 30 ms of voice (240 PCM samples) are packetized. This packetization results in a lower packet rate of 33.3 pps. For the 60 ms of voice shown in the figure, only two packets (carrying 30 ms of voice each) are generated, so the IP overhead is reduced by one third compared to the first scenario. The default value for the packetization period on most Cisco VoIP devices is 20 ms. This default is the optimal value for most scenarios. When you consider increasing this value to benefit from lower IP encapsulation overhead, you also have to consider that a higher packetization period causes a higher delay. The extra delay is introduced during packetization because more voice information has to be collected before a packet can be generated and sent. Caution
You should only increase the default packetization period if you are sure that the additional delay can be accepted (for instance, if other sources of delay, such as buffering, are rather small) and if you cannot solve bandwidth issues by any other means (for example, using cRTP or adding bandwidth). Because of the additional overhead, you should avoid trying to reduce delay by decreasing the default packetization period and use other methods to reduce delay (such as QoS or adding bandwidth). The table in Figure shows examples of VoIP encapsulation with varying codecs and packetization periods. You can see that an increased packetization period increases the size of the IP packet while reducing the packet rate. In the table, the IP overhead is assumed to be 40 bytes. This value is the normal value for VoIP packets composed of 20 bytes of IP header, 8 bytes of UDP header, and 12 bytes of RTP header. If cRTP is used, the IP overhead decreases. Note
The table in Figure shows the IP packet size only and does not consider the overhead that is caused by data-link encapsulation.
Content 2.4 Calculating Bandwidth Requirements for VoIP 2.4.4 Data-Link Overhead When a VoIP device sends IP packets over a link within an IP network, the device encapsulates the packets using the data-link protocol for that link. Each link can use a different data-link protocol. Figure illustrates an IP packet that is going from one IP phone to another. The two IP phones are located in different LANs and separated by a Frame Relay network. Before the sending telephone transmits the VoIP packet onto the LAN, the telephone has to encapsulate the packet into an Ethernet frame. The router that receives the frame removes the Ethernet header and encapsulates the VoIP packet into Frame Relay before sending the packet out to the WAN. The router receiving the VoIP packet from the Frame Relay network removes the Frame Relay header and encapsulates the VoIP packet into an Ethernet frame again before passing the packet on to the receiving IP phone. As illustrated in Figure , the Ethernet header and the Frame Relay header differ in size. The overhead of data-link protocols commonly used for VoIP is 18 bytes for Ethernet, 22 bytes for 802.1Q-tagged Ethernet frames, and 6 bytes for Frame Relay or multilink PPP (MLP). When you calculate the bandwidth