Voice Over Internet Protocol Voip) Recommendation for Technical Manual and Guidance Material

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.

2 ACP-WGI-2/WP-211 15 August 2007

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. 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.

User Interface OSI- Layers

Layer 7, 6, and 5 T.120 H.450.1 H.323 H.235 (RTP)

T.130 H.225 H.225 H.245 SIP (AVC) 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 Users

Figure 1. VoIP Architecture & Layers

2. References

1 ANC Action Report No.379, Consolidation of the Work of Panels February 11, 2003 2 AN-WP/7809, February 2003 Approval Of An Executive Summary For A New Task And Of The Establishment Of A New Study Group 3 AN-WP/7820, February 28, 2003 Review Of The Report Of The AMCP/8 Meeting On Agenda Item 7 (Future Work) 4 ICAO-Doc 9804, AN/762, First Manual on Air Traffic Services (ATS) Edition-2002 Ground-Ground Voice Switching and Signalling 5 EUROCAE, EUR053-04/GT67-2: Minutes of 1st Meeting of Working March 15, 2004 Group 67 (VoIP for ATM) 6 ITU H.323 version 5: July 2003 Packet-based multimedia communications systems

7 ITU H.225.0: July 2003 Call Signalling Protocols and media stream packetization for packet-based multimedia communication systems 8 ITU H.235: August 2003 Security and encryption for H-series (H.323 and other H.245-based) multimedia terminals 9 ITU H.245: July 2003 Control Protocol for multimedia communication 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 ISDN user-network interface layer 3 Amendment 1: December 2002 specification for basic call control. Extensions for the support of digital multiplexing equipment. 14 ITU H.450.1: February 1998 Generic functional protocol for the support of supplementary services in H.323 15 ITU-T.120: July 1996 and Annex C, Data protocols for multimedia February 1998 conferencing 16 ITU-T.130: February 1998 Audio Video and Control for Conferences Multimedia Architecture/General Vision

17 ITU-T G.711 November 1988, Pulse code modulation (PCM) of voice Appendixes I and II frequencies 18 ITU-T G.728 September 1992 Coding of speech at 16 kbps suing low- delay code excited linear prediction 19 ITU-T G.729 March 1996 Coding of speech at 8 kbps using conjugate-structure algebraic-code- excited linear-prediction (CS-ACELP) 20 IETF RFC 3261, June 2002 SIP: Session Initiation Protocol 21 IETF RFC 3262, June 2002 Reliability of Provisional Responses in the Session Initiation Protocol (SIP) 22 IETF RFC 3263, June 2002 Session Initiation Protocol (SIP): Locating SIP Servers 23 IETF RFC 3264, June 2002 An Offer/Answer Model with the Session Description Protocol (SDP) 24 IETF RFC 3265, June 2002 Session Initiation Protocol (SIP)-Specific Event Notification 25 IETF RFC 3550, July 2003 RTP: A Transport Protocol for Real-Time Applications 26 IETF RFC 2326, April 1998 Real Time Streaming Protocol (RTSP) 27 IETF RFC 3525, June 2003 Gateway Control Protocol Version 1 28 ECMA 312, 3rd edition, June 2003 Private Integrated Services Network (PISN) – Profile Standard for the Use of PSS1 (QSIG) in Air Traffic Services Networks 29 IETF RFC 791: 1981 Internet Protocol Specification 30 IETF RFC 793: 1981 Transmission Control Protocol 31 IETF RFC 768: 1981 User Data-gram Protocol 32 IETF RFC 2460: 1998 Internet Protocol, Version 6 (IPv6) Specification 33 IETF RFC 2327, June 2002 SDP: Session Description Protocol 34 IETF RFC 3266, April 1998 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.

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 real- time, 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.

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 SW Network

PSTNPSTN

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)].

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 Signalling

Audio Video H.450.1 Series Codec Codec T.120 (Supplementary s s (Real Services) Time) G.711 H.261 G.728 H.263 RTCP T.130 G.729 (Real (Audio- H.225.0 H.235 Time Visual RAS (Security Transpor Control) Q.931 ) tControl (Call H.245 Protocol) Signallin (Control RTP g) Signallin g) UDP (User Datagram TCP (Transfer Control Protocol) Protocol) IP (Internet Protocol) v4 or v6

Figure 3. H.323 Architecture

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).

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

AV I/O PINT Data Application and system control Services equipment (Interface e.g., to PSTN using Call Transfer/Conferences/Call Signalling RTSP Hold/ Call Monitoring and other Audio Video e.g. SS7, Supplementary Services ATS- QSIG) Extension Methods Headers SIP

Message Body (e.g. SDP) RTP

TCP UDP IP

Figure 4: SIP Protocol Suite

MEGACO Gateway

IP IP IP PSTN Network NetworkNetwork

Figure 5. VoIP interface to PSTN via MEGACO

SIP Server SIP/TCP or UDP

IPIP IPIP NetworkNetwork NetworkNetwork

IPIP RTP/UDP NetworkNetwork

Figure 6. VoIP with SIP

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.

8. Glossary

A-G Air to Ground ATM Air Traffic Management, Asynchronous Transfer Mode ATO-P Air Traffic Organization-Planning ATS Air Traffic Services AV Audio Video AVC Audio-visual Control AVSSSG ATS Voice Switching Signalling Study Group BGP Border Gateway Protocol CODEC coder/decoder DTMF Dual Tone Multifrequency E-1 European digital signalling level-1 ETSI European Telecommunications Standards Institute EUROCAE European Organization for Civil Aviation Equipment G-G Ground to Ground ICAO International Civil Aviation Organization IETF Internet Engineering Task Force IP Internet Protocol IPv4/IPv6 Internet Protocol version 4 and 6 IPSec Internet Protocol Security ISDN Integrated Services Digital Network ITU-T International Telecommunications Union (telecommunications sector) LAN Local Area Network MEGACO Media Gateway Control MFC-R2 MultiFrequency Compelled R2 (analog signalling standard) MGCP Media Gateway Control Protocol MPLS Multiprotocol Label Switching NASA National Aeronautics and Space Administration OSPF Open Shortest Path First Protocol PINT PSTN/Internet Networking PISN Private Integrated Services Network PSTN Private Switched Telephone Network QoS Quality of Service QSIG Q-signalling RAS Registration, Administration, Status RTP Real-time Transport Protocol RTCP Real-time Transport Control Protocol SDP Session Description Protocol SIP Session Initiation Protocol SSL Secure Sockets Layer T-1 North American digital signalling level-1 TCP Transport Control Protocol TLS Transport Layer Security Type 5/7/9 FAA proprietary analog communication signalling protocols for inbound/outbound/address, DTMF, Voice Call

UDP User Datagram Protocol VoIP Voice Over Internet Protocol VPN Virtual Private Network WAN Wide Area Network