ACP-WGI-2/WP-211
15 August 2007
AERONAUTICAL COMMUNICATIONS PANEL (ACP)
Working Group I – Internetworking
Working Paper
Voice Over Internet Protocol (VoIP) Recommendation
For
Technical Manual and Guidance Material
Prepared by Kelly Kitchens
FAA (ATO-P)
August 2007
SUMMARY
This working paper proposes an approach to support the inclusion of voice over IP
services into the IPS Technical Manual and Guidance Material.
1
ACP-WGI-2/WP-211
15 August 2007
1. Introduction
This paper is a response to an action item to provide a recommendation and approach for voice over IP
(VoIP) support in the IPS technical manual and/or guidance material.
1.1 Background
Currently member states are using VoIP technology to support aviation air traffic services. In addition,
ICAO is developing standards for the implementation of IP networking services via the ATN. In order
to ensure that member states have the ability to deploy VoIP service over the ATN, it was determined
that the IPS Standards provide support for VoIP services.
1.2 Scope
This working paper focuses on approaches for implementing VoIP and IP telephony standards support
in either the technical manual or guidance material.
2. References
1
ACP-WG I-01/WP/14
VoIP Papers for Review
2
WP XXX
Implementation of VoIP in ATS
3. Discussion
Contained in the references are papers that address the subject of VoIP. The two significant papers are
the Implementation of VoIP in ATS (attached) and the Implementation of Voice over Internet Protocol
(VoIP) for Air Traffic Management (ATM) Applications, which is also known as the VoIP Handbook.
VoIP currently has two major approaches for deployment; H.323 and SIP. In either approach the
required protocols to implement VoIP service reside in the application layer. Therefore, since the
Manual on Detailed Technical Specification for the ATN covers the network layer and the VoIP
functionality are not implemented in the network, the need to include VoIP in the technical manual are
not necessary.
Figure 1
4. Recommendations
Therefore it is recommended that the VoIP Handbook and the attached paper be incorporated in to the
guidance document in order to allow member states to deploy VoIP service if required. However, there
is a need to ensure QoS issues are also addressed for VoIP service and the meeting participants are
invited to share ideas on this topic.
Finally, a need may also exist to bring awareness to member states that this information exist. This has
been done in the SARPs, and to be consistent, information or a note should be added to the technical
manual informing readers that VoIP support is contained in the guidance document.
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ACP-WGI-2/WP-211
15 August 2007
3
Attachment 1
WP No.______
AERONAUTICAL COMMUNICATIONS PANEL (ACP)
Working Group N – Networking
Working Paper
Implementation Of Voice Over Internet Protocol (VoIP)
In Air Traffic Services (ATS)
Prepared by Leon Sayadian
FAA (ATO-P)
May 2004
SUMMARY
This working paper proposes a concept, and presents an alternative for integrating digital
voice and data in the ATN Ground-to-Ground (G-G) Infrastructure using Voice over
Internet Protocol (VoIP) technology.
1
1. Introduction
The current Air Traffic Services (ATS) voice switches provide air traffic controllers with the capability
to establish Air-Ground (A-G) and Ground-Ground (G-G) voice communications. The current G-G
infrastructure uses dedicated analog lines to communicate between air traffic facilities. This
proprietary technology is becoming obsolete, inefficient and costly to maintain. Modern scalable
digital technology is mature, cost effective, and can adapt existing infrastructures to converge voice and
data using Voice Over Internet Protocol (VoIP) technology.
1.1 Background
ICAO has initiated a technical research effort to identify new technologies to replace the current analog
voice communications system [1,2,3]. The ATS Voice Switching and Signaling Study Group
(AVSSSG) was convened to update ICAO Annex 10 and 11 with provisions for digital technology and
subsequently issue an associated guidance document [4].
However, this document is not a complete guide to deploying G-G voice communications networks; in
addition current technologies are being outpaced by the rapid progress in telecommunications research
and development. Currently, various public and private sector entities (e.g., NASA, Boeing, Internet
Engineering Task Force (IETF), ETSI, EUROCONTROL, and EUROCAE) are working to develop
standardized VoIP services for Air Traffic Management (ATM) application [5].
1.2 Scope
This working paper focuses on approaches for implementing VoIP and IP telephony for the G-G analog
voice switching system using digital Commercial Off The Shelf (COTS) products, which are in
compliance with accepted standards and protocols. Voice over IP is based on Open System
Architecture model, as shown in Figure 1.
Issues regarding mobile IP and A-G applications are beyond the scope of this paper.
2
User Interface
OSI- Layers
Layer 7,
6, and 5
T.120
(RTP)
H.450.1
H.323
H.235
T.130
(AVC)
H.225
H.225
H.245
Q.931
RAS
Layer 4
TCP/UDP
Layer 3
IPv4 or IPv6
Layer 2
FR, ATM, ATS-QSIG, etc.
Layer 1
Physical Interfaces T1/E1
Users
LAN/WAN/PSTN
SIP
Users
Figure 1. VoIP Architecture & Layers
3
2. References
1
2
ANC Action Report No.379,
February 11, 2003
AN-WP/7809, February 2003
3
AN-WP/7820, February 28, 2003
4
ICAO-Doc 9804, AN/762, First
Edition-2002
5
6
EUROCAE, EUR053-04/GT67-2:
March 15, 2004
ITU H.323 version 5: July 2003
7
ITU H.225.0: July 2003
8
ITU H.235: August 2003
9
ITU H.245: July 2003
10
ITU H.261: March 1993
Video Codec for Audiovisual services
11
ITU H.263: February 1998
Video Coding for Low Bit Rate
Communication
12
ITU H.248: June 2000
Gateway Control Protocol
13
ITU Q.931: May 1998, with
Amendment 1: December 2002
14
ITU H.450.1: February 1998
15
ITU-T.120: July 1996 and Annex C,
February 1998
ITU-T.130: February 1998
ISDN user-network interface layer 3
specification for basic call control.
Extensions for the support of digital
multiplexing equipment.
Generic functional protocol for the
support of supplementary services
in H.323
Data protocols for multimedia
conferencing
Audio Video and Control for Conferences
Multimedia Architecture/General Vision
16
Consolidation of the Work of Panels
Approval Of An Executive Summary For
A New Task And Of The Establishment
Of A New Study Group
Review Of The Report Of The AMCP/8
Meeting On Agenda Item 7 (Future
Work)
Manual on Air Traffic Services (ATS)
Ground-Ground Voice Switching and
Signalling
Minutes of 1st Meeting of Working
Group 67 (VoIP for ATM)
Packet-based multimedia communications
systems
Call Signalling Protocols and media
stream packetization for packet-based
multimedia communication systems
Security and encryption for H-series
(H.323 and other H.245-based)
multimedia terminals
Control Protocol for multimedia
communication
4
17
18
ITU-T G.711 November 1988,
Appendixes I and II
ITU-T G.728 September 1992
19
ITU-T G.729 March 1996
20
21
IETF RFC 3261, June 2002
IETF RFC 3262, June 2002
22
IETF RFC 3263, June 2002
23
IETF RFC 3264, June 2002
24
IETF RFC 3265, June 2002
25
IETF RFC 3550, July 2003
26
27
28
IETF RFC 2326, April 1998
IETF RFC 3525, June 2003
ECMA 312, 3rd edition, June 2003
29
30
31
32
IETF RFC 791: 1981
IETF RFC 793: 1981
IETF RFC 768: 1981
IETF RFC 2460: 1998
33
34
IETF RFC 2327, June 2002
IETF RFC 3266, April 1998
Pulse code modulation (PCM) of voice
frequencies
Coding of speech at 16 kbps suing lowdelay code excited linear prediction
Coding of speech at 8 kbps using
conjugate-structure algebraic-codeexcited linear-prediction (CS-ACELP)
SIP: Session Initiation Protocol
Reliability of Provisional Responses in
the Session Initiation Protocol (SIP)
Session Initiation Protocol (SIP):
Locating SIP Servers
An Offer/Answer Model with the Session
Description Protocol (SDP)
Session Initiation Protocol (SIP)-Specific
Event Notification
RTP: A Transport Protocol for Real-Time
Applications
Real Time Streaming Protocol (RTSP)
Gateway Control Protocol Version 1
Private Integrated Services Network
(PISN) – Profile Standard for the Use of
PSS1 (QSIG) in Air Traffic Services
Networks
Internet Protocol Specification
Transmission Control Protocol
User Data-gram Protocol
Internet Protocol, Version 6 (IPv6)
Specification
SDP: Session Description Protocol
Support for IPv6 in SDP
3. Assumptions






A robust IP infrastructure exists that supports ATM requirements (e.g., availability,
performance, Quality of Services (QoS), security) at ATS facilities.
Interfaces are available to the Private Switched Telephone Network (PSTN) for backup and load
sharing.
The IP infrastructure is compatible with the legacy end systems (e.g., voice switches, circuits,
signalling protocols).
Member states manage the network segment within their domain.
Provisions are available for fixed wireless links (e.g., satellite).
ATS-QSIG signalling is integrated within the network.
5
4. Discussion
The ICAO initiative to migrate ATN towards new technologies (e.g., TCP/UDP/IPv4 or IPv6-based
architecture) opens up opportunities for implementing cost-effective technologies for the ATS. In
particular, ATN stakeholders now have the means to provide scalable, available, and economical
G-G communications among ATS facilities across intra- and inter-domains.
Currently, ATS voice communications infrastructures must contend with the burden of maintaining
costly and congested point-to-point trunk circuits that are dedicated to a particular services and
capability. This infrastructure also requires proprietary signalling protocols that are difficult to
maintain (e.g., MFC-R2, Type 5/7/9).
The implementation of VoIP to provide voice, data, and signalling services over a ubiquitous TCP/IP
[29, 30] protocol stack will achieve cost-effective for leveraging a shared medium for these payloads.
These are ported over the data-centric IP infrastructure by digitizing, compressing, and converting
voice and video into packets. These packets are transmitted over the network, along with the data and
signalling packets. Signalling protocols are used to set up and tear down calls, and convey information
for locating users and negotiating network services.
4.1 Performance Criteria and Mechanisms
This section will present some considerations for implementing VoIP technology in order to maximize
QoS and performance. Performance criteria will be described, and appropriate mechanisms will be
discussed for managing these parameters for G-G ATM communications.
4.1.1 Quality of Voice
Since IP was initially designed for data, mechanisms have been implemented to provide for the realtime, low-latency, and error-correction demands for voice. These mechanisms include:






Echo cancellation
Packet prioritization – prioritizes voice packets over other traffic
Forward Error Correction
Low Delay CODEC
Bandwidth allocation and queuing
Network delay and jitter buffering
4.1.2 Interoperability
To ensure compatibility among disparate vendor product lines, the H.323 framework may be
implemented as a common standard for voice and data communications over packet networks (e.g.,
IPv4 or IPv6 [29,32]), as shown in Figure 1.
6
4.1.3 Security
VoIP security is implemented with H.235, designed for multimedia end systems at the application
level. Security at the transport layer [e.g. Secure Sockets Layer (SSL), Transport Layer
Security (TLS)], network layer (e.g. IPSec), and link layer [e.g. Virtual Private Network (VPN) and
Multiprotocol Label Switching (MPLS)] will also be provided.
4.1.4 Integration with Private Switched Telephone Network (PSTN)
For a more robust architecture that provides back up and load sharing services for VoIP, an interface to
the legacy PSTN may be implemented with the H.248 Media Gateway Control Protocol (MGCP) [12],
as shown in Figure 2.
IPIP
Network
Network
SW
SW
GW
GW
PSTN
PSTN
LAN
LAN
Figure 2. Integration of PSTN Backup with VoIP
LEGEND
GW Gateway
LAN Local Area Network
SW MGCP Switch
4.1.5 Scalability
Expected growth in ATS communication paths will be accommodated in the VoIP infrastructure
through the use of intra-domain routing protocols [e.g., Open Shortest Path First (OSPF)] and
interdomain routing protocols [e.g., Border Gateway Protocol Version 4 (BGP-4)].
7
4.2 Standards and Protocols
There are two standardized frame works for implementing VoIP, H.323 and SIP. Although both
protocols may be used for VoIP applications, the original focus of each protocol is different. The focus
of H.323 has been to handle voice and multimedia calls, including supplementary services, while SIP
was designed as a generic transaction protocol for session initiation not bound to any specific media
(e.g., audio or video). Details of relevant protocols are described in the following subsection:
4.2.1
H.323 Packet-based multimedia communications systems
As shown in figure 1, H.323 [6] operations in the application layer to support multimedia protocols.
Figure 3 depicts the various protocols used to convey multimedia traffic over TCP/UDP/IP networks.
H.323 Core
Multimedia
Data Transfer
Audio
Codecs
Video
Codecs
G.711
G.728
G.729
H.261
H.263
Signalling
H.450.1 Series
(Supplementary
Services)
T.120
(Real
Time)
RTCP
(Real Time
Transport
Control
Protocol)
T.130
(AudioVisual
Control)
RTP
UDP (User Datagram Protocol)
H.225.0
RAS
Q.931
(Call
Signalling)
H.235
(Security)
H.245
(Control
Signalling)
TCP (Transfer Control Protocol)
IP (Internet Protocol) v4 or v6
Figure 3. H.323 Architecture
8
4.2.2
Multimedia
This group of protocols converts between analog (e.g., voice) and digital signals, which are fed into, or
picked from, the UDP/IP network. Some of these protocols include:
 Audio codecs – These compress digital voice for low bandwidth transmission, and decompress
digital voice received from the network for feeding to the user audio device (e.g., speaker,
headphone) [17, 18, 19].
 Video codecs – These compress digital video for constrained bandwidth, and decompress digital
video received from the network for feeding to the user video device [10, 11].
 RTP (Real-Time Transport Protocol) and RTP Control Protocol (RTCP) [25] – These are
control protocols for the payloads fed into the network. RTP regulates the end-to-end delivery
of audio and video in real time over IP networks. RTCP regulates the control services in
multimedia transmissions, and monitors the quality of its distribution, including synchronization
of receivers.
4.2.3
Data Transfer
This class of protocols provides real-time, multi-point data communications and application services
over IP networks (e.g., collaborative decision making with video, voice, and data exchange). Data
transfers between generic applications and the IP network are processed by the T.120 protocol [15],
which can operate over various transports, including PSTN and ISDN.
T.130 [16] is a protocol still under development for controlling audiovisual sessions for real-time
multimedia conferencing, and ensure high QoS.
4.2.4
Signalling
H.225.0 [7] call signalling is used to set up connections and exchange call signalling between H.323
endpoints (terminals and gateways), which are transported as real-time data and carried over the
TCP/UDP/IP network. H.225.0 uses Q.931 [13] for call setup and teardown.
H.245 [9] control signalling is used to exchange end-to-end messages between H.323 endpoints. The
control messages are carried over H.245 logical control channels, which are relayed between
conference session endpoints.
H.235 [8] provides security services within the H.323 framework, such as authentication, encryption,
integrity and no-repudiation.
H.450.1 [14] deals with the procedures and signalling protocol between H.323 entities for the control of
supplementary services. Other protocols within the H.450 series (i.e. H.450.2-12) provide specific
supplementary services (e.g., call transfer, call hold, call waiting, call priority).
9
5. IP and MGCP/MEGACO
As shown in Figure 1, SIP [20, 21, 22, 24] is an application layer control protocol that provides
advanced signalling and control functionality for large range multimedia communications. SIP is an
alternative to H.323, which establishes, modifies, and terminates multimedia sessions, which can be
used for IP telephony. SIP is an important component in the context of other protocols to enable
complete multimedia architecture, as shown in Figure 4. These include RTP [25] for real time data
transport and QoS assurance, RTSP [26] for controlling streaming media, MEGACO [27] for
controlling gateways to the PSTN (see Figure 5), and SDP [23] for describing multimedia sessions.
These sessions include Internet multimedia conferences, Internet telephone calls, and multimedia
distribution over TCP/UDP/IPv4, or IPv6, as shown in Figure 6.
SIP Suite
PINT
(Interface
to PSTN
Signalling
e.g. SS7,
ATSQSIG)
Data
Services
e.g.,
using
RTSP
Application and system control
Call Transfer/Conferences/Call
Hold/ Call Monitoring and other
Supplementary Services
Extension
Headers
SIP
Audio
Video
Methods
Message Body (e.g. SDP)
TCP
AV I/O
equipment
RTP
UDP
IP
Figure 4: SIP Protocol Suite
10
MEGACO
Gateway
IP
IP
IP
PSTN
Network
Network
Network
Figure 5. VoIP interface to PSTN via MEGACO
SIP
Server
SIP/TCP or UDP
IP
IP
Network
Network
IP
IP
Network
Network
IP
IP
Network
Network
RTP/UDP
Figure 6. VoIP with SIP
11
6. Advantages of voice over data network
The key advantages associated with the use of a packet network for the transmission of digitized
voice are:






Bandwidth allocation efficiency
Ability to use modern voice compression methods
Associate economics with shared network use
Reduce costs
Enhanced reliability of packet networks
Ability to use multiple logical connections over a single physical circuit.
7. Recommendations
VoIP implementation for future ATM communications is recommended as an enhancement to current
switching capabilities by providing a dynamic routing function of increasing availability of the
communications infrastructure. Once the assumptions of Section 3 are satisfied, implementing
communications control in the network layer will produce cost-effective management and maintenance
through shared media for voice and data. It is recommended that the ICAO/ACP Working Group “N”
support the adoption of VoIP by EUROCAE Working Group 67 in the ATM G-G community to
implement this concept for ATS. Upon approval of the VoIP technology, the AVSSSG Guidance
Manual should be updated.
12
8. Glossary
A-G
ATM
ATO-P
ATS
AV
AVC
AVSSSG
BGP
CODEC
DTMF
E-1
ETSI
EUROCAE
G-G
ICAO
IETF
IP
IPv4/IPv6
IPSec
ISDN
ITU-T
LAN
MEGACO
MFC-R2
MGCP
MPLS
NASA
OSPF
PINT
PISN
PSTN
QoS
QSIG
RAS
RTP
RTCP
SDP
SIP
SSL
T-1
TCP
TLS
Type 5/7/9
Air to Ground
Air Traffic Management, Asynchronous Transfer Mode
Air Traffic Organization-Planning
Air Traffic Services
Audio Video
Audio-visual Control
ATS Voice Switching Signalling Study Group
Border Gateway Protocol
coder/decoder
Dual Tone Multifrequency
European digital signalling level-1
European Telecommunications Standards Institute
European Organization for Civil Aviation Equipment
Ground to Ground
International Civil Aviation Organization
Internet Engineering Task Force
Internet Protocol
Internet Protocol version 4 and 6
Internet Protocol Security
Integrated Services Digital Network
International Telecommunications Union (telecommunications sector)
Local Area Network
Media Gateway Control
MultiFrequency Compelled R2 (analog signalling standard)
Media Gateway Control Protocol
Multiprotocol Label Switching
National Aeronautics and Space Administration
Open Shortest Path First Protocol
PSTN/Internet Networking
Private Integrated Services Network
Private Switched Telephone Network
Quality of Service
Q-signalling
Registration, Administration, Status
Real-time Transport Protocol
Real-time Transport Control Protocol
Session Description Protocol
Session Initiation Protocol
Secure Sockets Layer
North American digital signalling level-1
Transport Control Protocol
Transport Layer Security
FAA proprietary analog communication signalling protocols for
inbound/outbound/address, DTMF, Voice Call
13
UDP
VoIP
VPN
WAN
User Datagram Protocol
Voice Over Internet Protocol
Virtual Private Network
Wide Area Network
14