Impact of Voice Samples and Packet Size on Bandwidth
DSP coverts analog voice signal to digital voice signal using a particular codec. Based on the codec used, the DSP generates so many bits per second. The bits that are generated for 10 milliseconds (ms) of analog voice signal form one digital voice sample. The size of the digital voice sample depends on the codec used. Table 1-6 shows how the digital voice sample size changes based on the codec used. The number of voice bytes for two digital voice samples using different codecs is shown in the last column.
Table 1-6 Examples of Voice Payload Size Using Different Codecs
Table 1-6 Examples of Voice Payload Size Using Different Codecs
|
Codec: Bandwidth |
Size of Digital Voice Sample for 10 ms of Analog Voice in Bits |
Size of 10 ms Digitized Voice in Bytes |
Size of Two Digital Voice Samples (20 ms) |
|
G.711: 64 Kbps |
64,000 bps X 10/1000 sec = 640 bits |
80 bytes |
2 X 80 = 160 bytes |
|
G.726 r32: 32 Kbps |
32,000 bps X 10/1000 sec = 320 bits |
40 bytes |
2 X 40 = 80 bytes |
|
G.726 r24: 24 Kbps |
24,000 bps X 10/1000 sec = 240 bits |
30 bytes |
2 X 30 = 60 bytes |
|
G.726 r16: 16 Kbps |
16,000 bps X 10/1000 sec = 160 bits |
20 bytes |
2 X 20 = 40 bytes |
|
G.728: 16 Kbps |
16,000 bps X 10/1000 sec = 160 bits |
20 bytes |
2 X 20 = 40 bytes |
|
G.729: 8 Kbps |
8000 bps X 10/1000 sec = 80 bits |
10 bytes |
2 X 10 = 20 bytes |
The total size of a Layer 2 frame encapsulating a VoIP packet depends on the following factors:
■ Packet rate and packetization size—Packet rate, specified in packets per seconds (pps), is inversely proportional to packetization size, which is the amount of voice that is digitized and encapsulated in each IP packet. Packetization size is expressed in bytes and depends on the codec used and the amount of voice that is digitized. For example, if two 10-ms digitized voice samples (total of 20 ms voice) are encapsulated in each IP packet, the packet rate will be 1 over 0.020, or 50 packets per second (pps), and if G.711 is used, the packetization size will be 160 bytes. (See Table 1-6.)
■ IP overhead—IP overhead refers to the total number of bytes in the RTP, UDP, and IP headers. With no RTP header compression, the IP overhead is 40 bytes. If cRTP with no header checksum is applied to a link, the IP overhead drops to 2 bytes, and with header checksum, the IP header checksum is 4 bytes.
■ Data link overhead—Data link layer overhead is always present, but its size depends on the type of encapsulation (frame type) and whether link compression applied. For instance, the data link layer overhead of Ethernet is 18 bytes (it is 22 bytes with 802.1Q).
■ Tunneling overhead—Tunneling overhead is only present if some type of tunneling is used. Generic routing encapsulation (GRE), Layer 2 Tunneling Protocol (L2TP), IP security (IPsec), QinQ (802.1Q), and Multiprotocol Label Switching (MPLS) are common tunneling techniques with their own usage reasons and benefits. Each tunneling approach adds a specific number of overhead bytes to the frame.
Codecs are of various types. The size of each VoIP packet depends on the codec type used and the
number of voice samples encapsulated in each IP packet. The number of bits per second that each
codec generates is referred to as codec bandwidth. The following is a list of some ITU codec
standards, along with a brief description for each:
■ G.711 is PCM—Based on the 8000 samples per second rate and 8 bits per sample, PCM generates 64,000 bits per second, or 64 Kbps. No compression is performed.
■ G.726 is adaptive differential pulse code modulation (ADPCM)—Instead of constantly sending 8 bits per sample, fewer bits per sample, which only describe the change from the previous sample, are sent. If the number of bits (that describe the change) sent is 4, 3, or 2, G.726 generates 32 Kbps, 24 Kbps, or 16 Kbps respectively, and it is correspondingly called G.726 r32, G.726 r24, or G.726 r16.
■ G.722 is wideband speech encoding standard—G.722 divides the input signal into two subbands and encodes each subband using a modified version of ADPCM. G.722 supports a bit rate of 64 Kbps, 56 Kbps, or 48 Kbps.
■ G.728 is low delay code exited linear prediction (LDCELP)—G.728 uses codes that describe voice samples generated by human vocal cords, and it utilizes a prediction technique. Wave shapes of five samples (equivalent of 40 bits in PCM) are expressed with 10-bit codes; therefore, the G.728 bandwidth drops to 16 Kbps.
■ G.729 is conjugate structure algebraic code exited linear prediction (CS-ACELP)—
G.729 also uses codes from a code book; however, 10 samples (equivalent of 80 PCM bits) are expressed with 10-bit codes. Therefore, the G.729 is only 8 Kbps.
DSPs produce one digital voice sample for 10 milliseconds (ms) of analog voice signal. It is common among Cisco voice-enabled devices to put two digital voice samples in one IP packet, but it is possible to put three or four samples in one IP packet if desired. The packetization period is the amount of analog voice signal (expressed in milliseconds) that is encapsulated in each IP packet (in digitized format). The merit of more voice samples in a packet—longer packetization period, in other words—is reduction in the overhead-to-payload ratio.
The problem, though, with putting too many digital voice samples in one IP packet is that when a packet is dropped, too much voice is lost. That loss has a more noticeable negative effect on the quality of the call when packets are dropped. The other drawback of a longer packetization period (more than two or three digital voice samples in one IP packet) is the extra packetization delay it introduces. More voice bits means a larger IP packet, and a larger IP packet means a longer packetization period.
Table 1-7 shows a few examples to demonstrate the combined effect of codec used and packet-ization period (number of digitized 10-ms voice samples per packet) on the voice encapsulating IP packet (VoIP) size and on the packet rate. The examples in Table 1-7 do not use compressed RTP and make no reference to the effects of Layer 2 and tunneling overheads.
Table 1-7 Packet Size and Packet Rate Variation Examples
Table 1-7 Packet Size and Packet Rate Variation Examples
|
Codec and Packetization Period (Number of Encapsulated Digital Voice Samples) |
Codec Bandwidth |
Voice Payload (Packetization) Size |
IP Overhead |
Total IP (VoIP) Packet Size |
Packet Rate (pps) |
|
G.711 with 20-ms packetization period (two 10-ms samples) |
64 Kbps |
160 bytes |
40 bytes |
200 bytes |
50 pps |
|
G.711 with 30-ms packetization period (three 10-ms samples) |
64 Kbps |
240 bytes |
40 bytes |
280 bytes |
33.33 pps |
|
G.729 with 20 ms packetization period (two 10-ms samples) |
8 Kbps |
20 bytes |
40 bytes |
60 bytes |
50 pps |
|
G.729 with 40 ms packetization period (four 10-ms samples) |
8 Kbps |
40 bytes |
40 bytes |
80 bytes |
25 pps |
Continue reading here: Data Link Overhead
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