Voice Delay Considerations

Voice call quality suffers when too much delay occurs. The symptoms include choppy voice, and even dropped calls. Interactivity also becomes difficult—ever had a call on a wireless phone, when you felt like you were talking on a radio? "Hey Fred, let's go bowling—OVER"— "Okay, Barney, let's go while Betty and Wilma are out shopping—OVER." With large delays, it sometimes becomes difficult to know when it is your turn to talk.

Voice traffic experiences delays just like any other packet, and that delay originates from several other sources. For a quick review on delay components covered so far, consider the delay components listed in Table 1-16.

Table 1-16 Components of Delay Not Specific to One Type of Traffic

Delay

Component

Definition

Where It Occurs

Serialization delay

Time taken to place all bits of a frame onto the physical medium. Function of frame size and physical link speed.

Outbound on every physical interface; typically negligible on T3 and faster links.

Propagation delay

Time taken for a single bit to traverse the physical medium from one end to the other. Based on the speed of light over that medium, and the length of the link.

Every physical link. Typically negligible on LAN links and shorter WAN links.

Queuing delay

Time spent in a queue awaiting the opportunity to be forwarded (output queuing), or awaiting a chance to cross the switch fabric (input queuing).

Possible on every output interface. Input queuing unlikely in routers, more likely in LAN switches.

continues continues

Table 1-16 Components of Delay Not Specific to One Type of Traffic (Continued)

Delay

Component

Definition

Where It Occurs

Forwarding or processing delay

Time required from receipt of the incoming frame until the frame/packet has been queued for transmission.

On every piece of switching equipment, including routers, LAN switches, Frame Relay switches, and ATM switches.

Shaping delay

Shaping (if configured) delays transmission of packets to avoid packet loss in the middle of a Frame Relay or ATM network.

Anywhere that shaping is configured, which is most likely on a router, when sending packets to a Frame Relay or ATM network.

Network delay

Delays created by the components of the carrier's network when using a service. For instance, the delay of a Frame Relay frame as it traverses the Frame Relay network.

Inside the service provider's network.

Figure 1-20 shows an example of delay concepts, with sample delay values shown. When the delay is negligible, the delay is just listed as zero.

Figure 1-20 Example Network with Various Delay Components Shown: Left-to-Right Directional Flow ->•

Delays for Packets Flowing Left-to-Right: Total Delay: 94 ms

Hannah

Forwarding: 0 Queuing: 0 Serialization: 0

Forwarding: 0 Queuing: 15 Serialization: 4 Propagation: .5

Forwarding: 0 Queuing: 0 Serialization: 0 Propagation: 0

Server 1

Forwarding: 0

Queuing: 15 Serialization: 9 Propagation: .5

Forwarding: 0 Queuing: 0 Serialization: 0 Propagation: 0

Server 1

Network: 50

(Note: Do Not Count R2 Serialization Here and at R2!)

Network: 50

(Note: Do Not Count R2 Serialization Here and at R2!)

The figure lists sample delay values. The values were all made up, but with some basis in reality. Forwarding delays are typically measured in microseconds, and become negligible. The propagation delay from R1 to R2 is calculated based on a 100-km link. The serialization delays shown were calculated for a G.729 call's packet, no compression, assuming PPP as the datalink protocol. The queuing delay varies greatly; the example value of 15 ms on R1's 56-kbps link was based on assuming a single 105-byte frame was enqueued ahead of the packet whose delay we are tracking—which is not a lot of queuing delay. The network delay of 50 ms was made up—but that is a very reasonable number. The total delay is only 94 ms—to data network engineers, the delay seems pretty good.

So is this good enough? How little delay does the voice call tolerate? The ITU defines what it claims to be a reasonable one-way delay budget. Cisco has a slightly different opinion. You also may have applications where the user tolerates large delays to save cash. Instead of paying $3 per minute for a quality call to a foreign country, for instance, you might be willing to tolerate poor quality if the call is free. Table 1-17 outlines the suggested delay budgets.

Table 1-17 One-Way Delay Budget Guidelines

1-Way Delay (in ms)

Description

0-150

ITU G.114's recommended acceptable range

0-200

Cisco's recommended acceptable range

150-400

ITU G.114's recommended range for degraded service

400+

ITU G.114's range of unacceptable delay in all cases

With the example in Figure 1-20, the voice call's delay fits inside the G.114 recommended delay budget. However, voice traffic introduces a few additional delay components, in addition to the delay factors that all data packets experience:

• Packetization delay

• De-jitter buffer delay (initial playout delay)

Be warned—many books and websites use different terms to refer to the component parts of these three voice-specific types of delay. The terms used in this book are consistent with the Cisco courses, and therefore with the exams.

Codec delay and packetization delay coincide with each other. To get the key concepts of both, consider Figure 1-21, which asks the question, "How much delay happens between when the human speaks, and when the IP packets are sent?"

Figure 1-21 Codec and Packetization Delays Between the Instant the Speaker Speaks and When the Packet Holding the Speech Is Sent

How Much Time Lags Between When Human Speaks and Packets Are Sent?

Packets Created by IP Phone

Packets Created by IP Phone

Consider what has to happen at the IP Phones before a packet can be sent. The caller dials digits, and the call is set up. When the call is set up, the IP Phone starts sending RTP packets. When these packets begin, they are sent every 20 ms (default)—in other words, each packet has 20 ms of voice inside the voice payload part of the packet. But how much time passes between when the speaker makes some sound and when the voice packet containing that sound is first sent?

Consider sounds waves for an instant. If you and I sit in the same room and talk, the delay from when you speak and when I hear it is very small, because your voice travels at the speed of sound, which is roughly 1000 km per hour. With packet telephony, the device that converts from sound to analog electrical signals, then to digital electrical signals, and then puts that digital signal (payload) into a packet, needs time to do the work. So there will be some delay between when the speaker speaks and when the IP/UDP/RTP payload packet is sent. In between when the speaker talks and when a packet is sent, the following delays are experienced:

• Packetization delay

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