The RTP protocol

The Real-Time Protocol (RTP) is responsible for the real-time transport of data such as audio and video. It was standardized in RFC 3550. It uses UDP as the transport protocol. To be transported, the audio or video has to be packetized by a codec. Basically, the protocol allows the specification of the timing and content requirements of the media transmission for the incoming and outgoing packets using the following:

• The sequence number
• Timestamps
• Packet forward without retransmission
• Source identification
• Content identification
• Synchronism

Codecs

A codec is an algorithm capable of encoding or decoding a digital stream. The content described in the RTP protocol is usually encoded by a codec. Each codec has a specific use. Some have compression while others do not. G.711 is still the most popular codec and does not use compression. With 64 Kbps of bandwidth for a single channel, it needs a high-speed network, commonly found in Local Area Networks (LANs). However, in Wide Area Networks (WANs), 64 Kbps can be too expensive to buy for a single channel. Codecs such as G.729 and GSM can compress the voice packets to as low as 8 Kbps, saving a lot of bandwidth. To simplify the way you choose a voice codec, the following table shows the most relevant ones. Bandwidths do not consider the lower protocol layer headers. There are also video codecs where the most relevant ones are the H.264 series and VP8 from Google.

There are other codecs such as G.723, GSM, iLBC, and SILK that are slowly losing ground to OPUS. OPUS is the codec adopted for the WebRTC standard. Obviously, you can dig a little more into codec details; there are dozens available, but I truly believe that the four described previously are the relevant choices at the time of this book being written. MOS is the Mean Opinion Score and defines the audio quality.