IP is primarily a unicast protocol. It was designed to move data from a single source to a single destination. IP, however, also defines multicast addresses. These are destination addresses that represent more than one destination device. The Internet Group Management Protocol (IGMP) manages multicast data flows.

With multicast, a single source sends data to multiple destinations at a single time. Each broadcast TV channel would have a unique IP multicast group, for example. Using IGMP protocol, clients can receive the broadcast packets and enable the routing of the broadcast stream to their network device through the network. While multicast saves on network bandwidth, there is unfortunately no reliability mechanism, so any lost packets stay lost.

Figure 1 shows a broadband network with three homes currently watching the same broadcast video stream of the 2008 Super Bowl. Each home, therefore, is part of an active multicast session, each receiving the same video bit stream, which originates from the IPTV headend.

Figure 1. Broadcast video using multicast in a broadband network.

Figure 2 shows a VOD unicast application. Data is sent from the VOD server to a single destination — in this case, a consumer’s home. For each unicast VOD session, there is a separate content stream on the network for each user. In this example, each stream might require 5MB/s of bandwidth for each user, requiring a total bandwidth of 15MB/s.

Figure 2. VOD using unicast in a broadband network.

If this is MPEG-2 HD video, it could require as much as 15Mb/s for each stream, or, in this example, a total of 45MB/s. Clearly, when using unicast protocol, network bandwidth requirements can quickly become huge.

Quality of service

The delivery of high-quality multimedia through a reliable network is core to IPTV. Multimedia applications feature various media types including text, graphics, animations, audio and video. There are many network-based multimedia applications today. Because of the importance of networked multimedia applications, it is critical for the IPTV network architect or content creator to understand the issues associated with this type of delivery system.

Within the network, multimedia data can be affected in the following ways: dropped packets, jitter in packet delivery times, delayed packets and data corruption. Even when the TCP protocol is used, the effectiveness of the IPTV service can be affected by the reliability and speed of the network.

The goal of Quality of Service (QoS) is to make sure the network can deliver end-to-end data with expected and predicted results. This includes latency, error rates, uptime, bandwidth and network traffic loads.

QoS is extremely important to a successful IPTV service. Only service operators that also own and manage the IP network to homes can guarantee a minimum QoS experience for customers. IPTV services that use the public Internet cannot guarantee the QoS necessary for a good user experience. To repeat, IPTV is not Internet video.

Real-time transport protocol

As noted earlier, IP networks were not designed for real-time delivery of data and can have unpredictable jitter and delay. To maintain QoS, the multimedia data that travels on the IP networks must arrive on time and in the same order it was sent. Real-time Transport Protocol (RTP) can be used to address the time-critical requirement of multimedia bit streams. RTP provides a timestamp and sequence number to IP packets containing media data. These can be used by the receiver to synchronize playback and manage buffers minimizing network jitter.

Encapsulating media data into IP packets

Delivering media bit streams over IP requires several layers of encapsulation. MPEG-2 transport streams, for example, consist of a series of 188-byte packets. These are grouped together and wrapped within an RTP packet. Finally, the RTP packet is encapsulated within a TCP or UDP datagram, forming an IP packet.

Figure 3 shows an RTP packet containing several MPEG-2 transport packets within its payload, all encapsulated using UDP in an IP packet. This diagram shows seven 188-byte transport packets that constitute the RTP payload. Each encapsulation adds additional header data and adds to the total required bandwidth.

Figure 3. An RTP packet encapsulated in UDP/IP.

If the network has sufficient QoS, it is possible to deliver media packets without the overhead of RTP. The packets are instead inserted directly into UDP packets. Figure 4 shows how MPEG-2 transport stream packets can be encapsulated within a UDP/IP packet.

Figure 4. MPEG-2 transport stream encapsulated in UDP/IP.

Sending MPEG-2 transport stream packets over UDP is used extensively within the private networks of cable and telephone companies to deliver MPEG-2 transport streams throughout the system. For general delivery over the unmanaged Internet without QoS guarantees, streaming protocols such as RTP may be used, but even then, packets may be lost in delivery, resulting in artifacts in the media presentation.

Delivering TV and movie services over IP promised to revolutionize almost every component of the TV industry. Just as the Internet changed how we ship, read the news and interact with others, television over IP could change how we integrate TV entertainment into our daily lives.

Combining the Internet and protocols can create an entirely new and exciting experience for viewers and new business opportunities for service providers. One key will be to maintain a high QoS across the entire network so viewers never suffer from the limitations we see in Internet video.