VOICE AND VIDEO OVER IP VOIP, RTP, RSVP

Can also handle voice and video TCP/IP Protocols Designed for data Can also handle voice and video

Representation Voice and video must be converted between analog and digital forms Typical device is codec (coder / decoder) Example encoding used by phone system is Pulse Code Modulation (PCM) Note: 128 second audio clip encoded in PCM requires one megabyte of memory Codec for voice, known as vocodec, attempts to recognize speech rather than just waveforms

Audio And Video Transmission And Reproduction Conventional isochronous architecture IP internet is not isochronous. Datagrams can be duplicated, delayed, or arrive out of order. Variance in delay is called jitter, and is especially pervasive in IP networks.

Illustration Of Jitter Buffer Data arrives in bursts Data leaves at steady rate

Real-Time Transport Protocol (RTP) Internet standard to transmit digitized audio or video signals

RTP provides two key facilities: A sequence number in each packet that allows a receiver to detect out-of-order delivery or loss a timestamp that allows a receiver to control playback.

RTP Message Format

RTP can coordinate multiple data streams Mixing RTP can coordinate multiple data streams Intended for combined audio and video Up to 15 sources Header specifies mixing

RTP is an application protocol RTP runs over UDP RTP Encapsulation RTP is an application protocol RTP runs over UDP

RTP Control Protocol (RTCP) Required part of RTP Allows sender and receiver to exchange information about sessions that are in progress Separate data stream Uses protocol port number one greater than port number of data stream

RTCP Message Types

a sender report message that provides an absolute timestamp. RTCP Message Types receiver report messages that inform the source about conditions of reception. a sender report message that provides an absolute timestamp. synchronize multiple streams.

RTCP Message Types Source description messages which provide general information about the user who owns or controls the source. canonical name for the stream owner, a character string in the form: user @ host

IP telephony or Voice over IP (VoIP) RTP used for encoding and transfer Also need signaling protocol for Dialing Answering a call Call forwarding Gateway used to connect IP telephone network to Public Switched Telephone Network (PSTN) PSTN uses SS7 for signaling

Standards For IP Telephony SIP (Session Initiation Protocol)

H.323 ITU standard Set of many protocols Major protocols specified by H.323 include

How H.323 Protocols Fit Together

Session Initiation Protocol (SIP) IETF standard Alternative to H.323 Less functionality Much smaller Permits SIP telephone to make call Does not require RTP for encoding

Session Description Protocol (SDP) Companion to SIP Specifies details such as Media encoding Protocol port numbers Multicast addresses