Cisco.IP.Telephony.Planning.Design.Implementation.Operation.and.Optimization [Electronic resources] نسخه متنی

Next-Generation Multiservice Networks

The desire of industry to combine distinct voice and data networks led to the development of several new concepts and technologies, such as packetized voice. Packetized voice comprises several standards and protocols. Applications use these protocols and standards to provide value-added and cost-effective services to users.

Packetized voice enables a device to send voice traffic (for example, telephone and fax) over an IP/Frame Relay/ATM network. In case of Voice over IP (VoIP), the digital signal processor (DSP) that is located on the voice gateway segments the voice signal into frames. The voice gateway combines these frames to form an IP packet and sends the packet over the IP network. On the receiving end, a reverse action converts the voice information that is stored in the IP packet into the original voice signal.

Across the IP network, these voice packets are transported by using the Real-Time Transport Protocol (RTP) and RTP Control Protocol (RTCP) stack and by using the User Datagram Protocol (UDP) as a transport layer protocol. RTP provides timestamps and sequence numbers in each packet to help synchronize the voice frames at the receiving side. RTCP provides a feedback mechanism that informs session participants of the received quality of the voice call and includes information such as delay, jitter, and lost packets.

It is important to note that most of the real-time applications use UDP as the transport layer protocol rather than TCP, for the following reasons:

TCP guarantees the retransmission of frames that are lost in the network, which is of no use in a packetized voice network because the late arrival of frames at the receiving end introduces delay. Hence, the capability of TCP to retransmit frames is not useful in packetized voice networks.

TCP introduces unnecessary delay by waiting for acknowledgements for every packet. This delay is not noticeable in the data networks but causes poor voice quality in packetized voice networks.

In addition to using RTP/UDP/IP as the protocol stack to carry voice calls across the IP network, VoIP networks use VoIP signaling protocols to set up and tear down the calls, carry the information to locate the users/phones, and exchange capabilities such as compression algorithms to be used during the conversation. The commonly used signaling protocols in VoIP networks are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), and Skinny Client Control Protocol (SCCP).

Introducing "packetized voice" capability into the router shown in Figure 1-1 turns the router into a voice gateway and enables the router to do the packetization just described along with the duties required as a regular data router. This change allows you to provide additional services such as toll bypass.

Toll bypass enables you to reduce the overall telephony expenditure by routing long-distance interoffice calls over existing packet-based WANs, thus avoiding interexchange carrier (IXC) toll charges. As shown in Figure 1-2, when you make phone calls between the enterprise headquarters and a branch location, you can send the voice calls via the WAN data circuit as opposed to using the Public Switched Telephone Network (PSTN). The connection from the PBX is now terminated on the gateway, allowing the gateway to receive both incoming and outgoing calls. This architecture also allows routing of a voice call over the internal network to the closest gateway to the destination of the call and then connects into the public network as a local phone call. This is called Tail-End Hop-Off (TEHO). You need to be aware that TEHO is not allowed in all countries. Hence, when you are designing a VoIP network with toll bypass or TEHO, you need to check the local government regulations.

To ensure that the design of your toll-bypass application is effective, it must include a voice gateway that is able to support multiple types of call signaling as it interacts with the PSTN, WAN, and existing PBX systems. By maintaining a connection to the PSTN, as shown in Figure 1-2, the gateway can reroute calls to the PSTN if congestion or failure occurs within the packet network.

Tip

By having this voice gateway in the network, you eliminate the need to have a separate "tie line" for the voice-only traffic.

The DSPs that are located on the voice gateways also have the capability to handle different types of compression algorithms, such as G.711, G.723.1, G.726, G.728, and G.729 traffic. These compression algorithms maximize the throughput on the packet network. Compression techniques on gateways, such as compressed RTP (cRTP), reduce the amount of bandwidth per voice call. You should be aware of the side effects of enabling cRTP on the voice gateways. cRTP processing on voice gateways essentially compresses all outgoing voice packets and decompresses the incoming compressed voice packets. Hence, if you have numerous voice calls across the WAN links, the router has to perform many cycles of this task, which can increase the amount of CPU resources consumed for cRTP, thus leaving less CPU cycles for the other tasks.

In traditional voice networks, each call consumes a fixed amount of bandwidth. The PBX does not place more calls than it can handle through the trunks connecting to the PSTN, as shown on the left side of Figure 1-3. In packetized networks, if bandwidth is available to make only two good-quality calls, in the absence of a call admission control (CAC) mechanism, the voice gateway allows the third call to go through, as shown on the right side of Figure 1-3. This third call degrades the quality of the existing two good voice calls. Hence, gatekeepers are deployed in the packetized networks to control the number of calls that can be sent over the WAN links. The CAC mechanism in the gatekeeper ensures that the gateway does not place the third call. Gatekeepers perform CAC and bandwidth management in VoIP networks. Gatekeepers ensure that enough bandwidth is available before granting permission to a gateway to place a call across the IP WAN. After receiving permission from the gatekeeper, the originating gateway initiates a call setup with the terminating gateway over the packet network.

Tip

When you are using IP networks to carry packetized voice traffic, an optional but important consideration is to use proper admission and bandwidth control mechanisms.

Besides performing CAC and bandwidth management, gatekeepers can perform accounting, call authorization, authentication (via RADIUS), address lookup and resolution, and translation between E.164 numbers and the IP addresses.