Media and Codecs

Although SIP meets the needs for signaling information in calls, VoIP PBXs still require a method to transmit the actual media stream, whether it is audio or video. The Real-Time Transport Protocol (RTP) is used in almost every VoIP implementation and was developed specifically for transmitting audio and video traffic across networks. Encryption of the media traffic was later added in the form of Secure Real-Time Transport Protocol (SRTP), which is what Lync Server 2013 uses by default to ensure that the media cannot be intercepted and played back.

It’s important to note that RTP and SRTP only provide a standard for carrying the media traffic which can be composed of various media codecs. Think of the RTP stream as a wrapper around the actual audio, which could be encoded by any type of codec. Media codecs are a way of translating audio and video data into bits that can be transmitted across a network. A codec analyzes an analog audio waveform and determines the best way to represent that audio stream as digital bits of 0s and 1s. That process is called encoding, and the reverse, decoding, is what the receiving end does in order to reconstruct the audio waveform for playback.

For two users to have an audio conversation, the codec used by both parties must match in order to correctly encode and decode the traffic. Figure 17.5 shows that while SRTP carries the real-time media, the parties must agree on a codec to use in order to actually have a conversation.

Figure 17.5. SIP signaling and SRTP media.