Voice Basics

Voice over data includes Voice over IP (VoIP), Voice over Frame Relay (VoFR), and Voice over ATM (VoATM). Each of these three voice over data technologies transports voice, and each is slightly different. Most of the questions you should see on an exam will be related to VoIP, and not VoFR or VoATM, because of the three options, VoIP is the most pervasive. Also calls between Cisco IP Phones use VoIP, not VoFR or VoATM.

Imagine a call between the two analog phones in Figure 1-17, extensions 201 and 301.

Figure 1-17 Call Between Analog Phones at Extensions 301 and 201

Server 1

Figure 1-17 Call Between Analog Phones at Extensions 301 and 201

Server 1

Before the voice can be heard at the other end of the call, several things must happen. Either user must pick up the phone and dial the digits. The router connected to the phone interprets the digits and uses signaling to set up the VoIP call. (Because both phones are plugged into FXS analog ports on R1 and R3, the routers use H.323 signaling.) At various points in the signaling process, the caller hears ringing, and the called party hears the phone ringing. The called party picks up the phone, and call setup is complete.

The actual voice call (as opposed to signaling) uses Real-Time Transport Protocol (RTP). Figure 1-18 outlines the format of an IP packet using RTP.

Figure 1-18 IP Packet for Voice Call: RTP

20 Bytes 8 Bytes 12 Bytes Variable




Voice Payload

Port Ra



nges: Popular 32767 G.711: 1 5orts) G729a:

Values: 60 Bytes 20 Bytes

In the call between the two analog phones, the router collects the analog voice, digitizes the voice, encodes the voice using a voice codec, and places the encoded voice into the payload field shown in Figure 1-18. For instance, R1 would create an IP packet as shown in Figure 1-18, place the encoded voice bits into the voice payload field, and send the packet. The source IP

address would be an IP address on R1, and the destination IP address would be an IP address on R3. When R3 receives the packet, it reverses the process, eventually playing the analog waveform for the voice out to the analog phone.

The IP Phones would experience a similar process in concept, although the details differ. The signaling process includes the use of Skinny Station Control Protocol (SSCP), with flows between each phone and the Cisco CallManager server. After signaling has completed, an RTP flow has been completed between the two phones. CallManager does not participate directly in the actual call, but only in call setup and teardown. (CallManager does maintain a TCP connection to each phone for control function support.) R1 and R3 do not play a role in the creation of the RTP packets on behalf of the IP Phone, because the IP Phones themselves create the packets. As far as R1 and R3 are concerned, the packets sent by the IP Phones are just IP packets.

Finally, the network administrator can choose from various coders/decoders (codecs) for the VoIP calls. Codecs process the incoming analog signal and convert the signal into a digital (binary) signal. The actual binary values used to represent the voice vary based on which codec is used. Each codec has various features, the most significant feature being the amount of bandwidth required to send the voice payload created by the codec. Table 1-11 lists the most popular codecs, and the bandwidth required for each.

Table 1-11 Popular Voice Codecs and Payload Bandwidth Requirements


Bit Rate for Payload* (in kbps)

Size of payload (20-ms Default in Cisco IOS Software)

G.711 Pulse Code Modulation (PCM)


160 bytes



80 bytes



20 bytes

G.723.1 ACELP


20 bytes**

* The payload contains the digitized voice, but does not include headers and trailers used to forward the voice traffic. ** G.723 defaults to a 30-ms payload per packet.

* The payload contains the digitized voice, but does not include headers and trailers used to forward the voice traffic. ** G.723 defaults to a 30-ms payload per packet.

This short section on voice basics (and yes, it is very basic!) can be summarized as follows:

• Various voice signaling protocols establish an RTP stream between the two phones, in response to the caller pressing digits on the phone.

• RTP streams transmit voice between the two phones (or between their VoIP gateways).

Why the relatively simple description of voice? All voice payload flows need the same QoS characteristics, and all voice signaling flows collectively need another set of QoS characteristics. While covering each QoS tool, this book suggests how to apply the tool to "voice"—for two subcategories, namely voice payload (RTP packets) and voice signaling. Table 1-12 contrasts the QoS requirements of voice payload and signaling flows.

Table 1-12 Comparing Voice Payload to Voice Signaling: QoS Requirements





Voice Payload





Voice Signaling





QoS tools can treat voice payload differently than they treat voice signaling. To do so, each QoS tool first classifies voice packets into one of these two categories. To classify, the QoS tool needs to be able to refer to a field in the packet that signifies that the packet is voice payload, voice signaling, or some other type of packet. Table 1-13 lists the various protocols used for signaling and for voice payload, defining documents, and identifying information.

Table 1-13 Voice Signaling and Payload Protocols


Documented By

Useful Classification Fields



Uses TCP port 1720



TCP ports 11xxx



TCP port 1720 (Fast Connect)

H.323/H.225 RAS


TCP port 1719



TCP ports 2000-2002

Simple Gateway Control Protocol (SGCP)

TCP ports 2000-2002

Media Gateway Control Protocol (MGCP)

RFC 2705

UDP port 2427, TCP port 2428

Intra-Cluster Communications Protocols (ICCP)


TCP ports 8001-8002

Real-Time Transport Protocol (RTP)

RFC 1889

UDP ports 16384-32767, even ports only

Real-Time Control Protocol (RTCP)

RFC 1889

UDP ports 16385-32767, odd ports only; uses RTP port + 1

The next few sections of this book examine voice more closely in relation to the four QoS characteristics: bandwidth, delay, jitter, and loss.

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