Voice over IP (VoIP)

David Feiner ACN 2004

Overview
Introduction VoIP & QoS H.323 SIP Comparison of H.323 and SIP Examples

Introduction
Voice Calls are transmitted over Packet Switched Network instead of Public Switched Telephone Networks (PSTN) Modes of Operation: - PC to PC - PC to Telephone - Telephone to PC - Telephone to Telephone

Figure: Cisco IP phones

Introduction - Why VoIP?

Cheaper calls Scalability Unified Messaging Mobility

VoIP & QoS

Voice quality characteristics
- Clarity: fidelity, clearness, and intelligibility of signal - Delay: effect on interactivity - Echo: distracting and confusing

Latency
- Components: Encoding, Packetization, Network delay, Receiver buffering, Decoding - ITU-TG.114 recommends 150ms
One-way Delay <100 -150ms 150 - 200ms Over 200 - 300ms Effect on perceived Quality Delay not detectable Acceptible quality; slight delay or hestitation noticeable Unacceptible delay; normal conversation impossible

VoIP & QoS (2)

Jitter
- Smoothed by playback buffers - Receivers adapt the depth of these buffers - Sudden changes in jitter may cause loss

Figure: Playback buffer

VoIP & QoS (3)

Bandwith
- Generally modest (64 kbps or less) - Depends on codec and use of silence suppression
Codec G.711 G.722 G.729 (A/B) Rate (kbps) 64 48-64 8

Packet loss
- Should be less then 5%

H.323
Recommendation published by ITU Ties together a number of protocols to allow multimedia transmission through an unreliable packet-based network 1996: approved by ITU 2003: Version 5

H.323 Architecture

H.323 Terminal Gateway Gatekeeper Multipoint Control Units (MCU)

H.323 Protocol Stack for VoIP

Speech G.7xx RTCP RTP UDP IP TCP H.225 Q.931 H.245 Control

G.7xx Speech (De)Coding

H.323 systems must support G.711: PCM, 64kbps Other codecs: G.729, G.726,

RTP
Realtime Transport Protocol (RFC 3550, July 2003) Application layer protocol for transmitting realtime data (audio, video, ...) Includes payload type identification, sequence numbering, timestamping, delivery monitoring Mostly over UDP Supports multicast & unicast

Control Protocol - RTCP

RTP Control Protocol (RFC 3550, July 2003) Periodic transmission of control packets to all participants in the session Main functions:

- provide feedback on quality of data distribution - carries a persistent transport-level identifier for an RTP source (CNAME) - each participant sends control packets to all others which independently observe the number of participants

More Control Protocols in H.323

H.225 (RAS)
- protocol between terminal and gatekeeper (if present) - allows terminals to join/leave zone, request/return bandwidth, provide status updates,

H.245 (Call Control)

- Media Control Protocol - Allows terminals to negotiate connection parameters (codec, bit rate, ..)

Q.931 (Call Signalling)

- Manages call setup and termination

SIP Session Initiation Protocol

Developed by IETF since 1999 RFC 2543, March 1999 (obsolete) RFC 3261, June 2002 Target: develop simpler and more modular protocol for VoIP than the large and complex H.323 by ITU

SIP (2)
SIP is a text-based protocol similar to HTTP and SMTP, for initiating interactive communication sessions between users SIP is an application-layer control (signalling) protocol for creating, modifying and terminating sessions with one or more participants Sessions include Internet Multimedia conferences, Internet Telephone calls and Multimedia distribution

SIP (3)
SIP can be used with different transport protocols, it doesn't even require reliable transport protocols A simple SIP client can be implemented using only UDP

SIP (4)

SIP (5)
UAC (user agent client) UAS (user agent server) SIP Terminal Caller application that initiates and sends SIP requests. Receives and responds to SIP requests on behalf of clients; accepts, redirects or refuses calls. Supports real-time, 2-way communication with another SIP entity. Supports both signalling and media, similar to H.323 terminal. Contains UAC. Contacts one or more clients or next-hop servers and passes the call requests further. Contains UAC and UAS. Accepts SIP requests, maps the address into zero or more new addresses and returns those addresses to the client. Does not initiate SIP requests or accept calls. Provides information about a caller's possible locations to redirect and proxy servers. May be co-located with a SIP server.

Proxy Server

Redirect Server

Location Server

Comparison of H.323 and SIP

Item Designed by Compatibility with PSTN Compatibility with Internet Architecture Completeness Parameter negotiation Call signaling Message format Media Transport H.323 ITU Yes No Monolithic Full protocol stack Yes Q.931 over TCP Binary RTP/RTCP SIP IETF Largely Yes Modular SIP just handles setup Yes SIP over TCP or UDP ASCII RTP/RTCP

Comparison of H.323 and SIP (2)

Item Multiparty calls Multimedia conferences Addressing Call termination Instant messaging Encryption Size of standards Implementation Status H.323 Yes Yes Host or tel. number No Yes 1400 pages Large and complex Widely deployed SIP Yes No URL Yes Yes 250 pages Moderate Up and coming

Explicit or TCP release Explicit or timeout

Examples
VONAGE
- founded in January 2001 - about 130.000 customers - www.vonage.com

Examples (2)
AT&T
Today, AT&T is rapidly evolving from a company that handles mostly long-distance voice calls to a company that provides data and voice communications over any distance www.att.com

Examples (3)
Inode
- G.729 used (13 kbps) - MPLS (Multi Protocol Label Switching) assures QoS - www.inode.at

Telekom Austria
- Offers IP Voice Services for companies - www.telekom.at

References
Larry L. Peterson / Bruce S. Davie; Computer Networks, A Systems Approach, 3rd Ed.; 2003 Andrew S. Tanenbaum; Computer Networks; 4th Ed.; 2003 RFC 3261 SIP: Session Initiation Protocol RFC 3550 RTP: A Transport Protocol for Real-Time Applications Packetizer A ressource for packet-switched conversational protocols, http://www.packetizer.com/ VoIP & QoS: You Cant Always Get What You Want http://people.internet2.edu/~ben/talks/tamu-voip-qos-4.02.pdf International Telecommunication Union, http://www.itu.int/ Cisco Systems, http://www.cisco.com/ AT&T, http://www.att.com/ VONAGE, http://www.vonage.com/ Inode, http://www.inode.at/ Telekom Austria, http://www.telekom.at/ Das deutschsprachige Voice over IP-Informationsportal, http://www.voip-info.de/