Media

Although SIP meets the needs for signaling information, VoIP PBXs still require a method to transmit the media stream. 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 uses by default to ensure that the media cannot be intercepted and played back.

SRTP only provides a standard for carrying the media traffic that can be of various media codecs. Media codecs are a way of translating audio and video data into bits that can be transmitted across a network. For two users to have an audio conversation, the codec used by both parties must match to correctly encode and decode the traffic. Although SRTP carries the real-time media, the parties must agree on a codec to have a conversation. Figure 17.5 displays this split of signaling and media traffic, which uses a specific codec such as RTAudio or G.711.

Figure 17.5 SIP Signaling and SRTP Media

Lync Server 2010 endpoints have the ability to use two different audio codecs. The default codec is Microsoft’s proprietary RTAudio codec, which can dynamically adjust its bandwidth to ensure a certain level of call quality. Lync endpoints can now also take advantage of the G.711 codec in certain scenarios that many VoIP implementations have used for years.

When Lync endpoints cannot communicate directly with another endpoint, the Mediation Server role can be used to transcode between RTAudio and G.711 codecs in a media stream. This is typical for when Lync endpoints communicate to a Mediation Server via RTAudio, but the Mediation Server may communicate with a media gateway via G.711. The Mediation Server acts as a translator in these scenarios.