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 Effect on perceived Quality
<100 -150ms Delay not detectable
150 - 200ms Acceptible quality; slight delay or hestitation noticeable
Over 200 - 300ms 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 Rate (kbps)
G.711 64
G.722 48-64
G.729 (A/B) 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 Control

G.7xx
RTCP H.225 Q.931 H.245
RTP

UDP TCP

IP
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 real-
time 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) Caller application that initiates and sends SIP requests.
UAS (user agent server) Receives and responds to SIP requests on behalf of
clients; accepts, redirects or refuses calls.
SIP Terminal Supports real-time, 2-way communication with another
SIP entity. Supports both signalling and media, similar
to H.323 terminal. Contains UAC.
Proxy Server Contacts one or more clients or next-hop servers and
passes the call requests further. Contains UAC and
UAS.
Redirect Server 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.
Location Server Provides information about a caller's possible locations
to redirect and proxy servers. May be co-located with a
SIP server.
Comparison of H.323 and SIP
Item H.323 SIP
Designed by ITU IETF
Compatibility with PSTN Yes Largely
Compatibility with Internet No Yes
Architecture Monolithic Modular
Completeness Full protocol stack SIP just handles setup
Parameter negotiation Yes Yes
Call signaling Q.931 over TCP SIP over TCP or UDP
Message format Binary ASCII
Media Transport RTP/RTCP RTP/RTCP
Comparison of H.323 and SIP (2)
Item H.323 SIP
Multiparty calls Yes Yes
Multimedia conferences Yes No
Addressing Host or tel. number URL
Call termination Explicit or TCP release Explicit or timeout
Instant messaging No Yes
Encryption Yes Yes
Size of standards 1400 pages 250 pages
Implementation Large and complex Moderate
Status Widely deployed Up and coming
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 Can’t 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/