Telephony Network Infrastructure using Session Initiation Protocol

1. Telephony Network Infrastructureusing Session Initiation Protocol Javier Martin-Perez Supervisor: Raimo Kantola Instructor: Marko Luoma 23-May-2012 Thesis work conducted at Comnet, Aalto University, Finland

2. Aims • VoIP (SIP) communications • Based telephony network (PSTN) • Call routing and admission control (CAC) for real time traffic flows • Priority usage perspective (MLPP)

3. Introduction • SIP to enable end-to-end voice calls • Multimedia communications in IP-based networks • voice/video calls over long distances • SIP global standard for voice communications • High cost of long-distance & intenational voice calls • Regulatory taxes imposed on long-distance • Not applicable to long-distance circuit for data traffic

4. Bigpicture MLPP, 3GPP SIP, SS7, 3GPP, H.323 Voice, video, text… European Commission Ficora SS7, SIGTRAN, TRIP CAC, BW brokers SBC, SRTP/SIPS

5. Network design

6. VoIP Core Network: SIP (1) • Text-based • Scalable • New functionalities added by extensions • Entities: • User Agent • Proxy • Register server • Register & location • GW/SBC used to interconnect to ISUP or H.323

7. VoIP Core Network: NAT traversal • Not supported by SIP • Solutions: • STUN, TURN & ICE • SBC

8. Session Border Controllers • Operates at both layer 3 and 5 • Full termination element (B2BUA) • Break end-to-end security • Modify certain SIP headers & message bodies • No route arithmetic (no shorter path for media packets) • Needs no modification on existing network • Bottle-neck (It handles signaling and media whereas SIP Proxies only signaling) • Hard to extend (If traffic increases  place a new or better one)

9. PSTN Network • Circuit-switched network • Termianls either analog or digital (ISDN) • SS7 (Signaling System 7) • Out-of-band signaling • ISDN • Terminals with digital access to PSTN

10. Interconnection ISUP-SIP • SIGTRAM: M3UA • MTP (layer 1-3, SS7) changes in IP node but ISUP remains the same • Between SG and MGC • Megaco(ITU-T H.248) • GW control protocol • Between MG and MGC • Alternative protocol MGCP SS7 node Signaling Gateway IP node • MG: converts circuit-switched voice to packet-based traffic

11. IMS Overlay Network • IP-based comm. system defined by the 3GPP • IMS accessed from: • 3G cellular system • Wireless LAN • Fixed networks • Planes architecture: • SDP carries media at user plane • Transport plane is packet-switched • Control plane managed by SIP signaling

12. Mobile Network • 3G Core Network connects to: • Packet-switched • Circuit-switched • 20% of the total traffic load in the network due to signaling traffic

13. Service Level Management • SLM: process managing the quality of services demanded by clients and offered by providers: • SLA: negotiated agreement between parties • QoS • Routing

14. SLM: QoS • IntServ: • To provide QoS to specific user packet streams • Admission control, classification, policing,queuing and scheduling • RSVP: reserves QoS for each flow in the network • Bandwidth reservation • DiffServ: scalable end-to-end model • Traffic differentiation • Call Admission Control (CAC) • Bandwidth Broker (BB): allocates network resources

15. SLM • Multilevel Precedence and Preemption: • Provides prioritized call handling service • Precedence involves assigning a priority level to a call • Preemption involves the seizing of resources. • Routing: Dynamic • IGP (Interior Gateway Protocol): Link state protocol • OSPF (RFC 2328), IS-IS (RFC 1142) • EGP (Exterior Gateway Protocol): Distance vector protocol • EGP (RFC 904), BGP (RFC 4271) • TRIP (RFC 3219): locates the optimum gateway out of a VoIP network into the PSTN.

16. Security • Confidentiality, integrity and scalability • TLS to protect SIP signaling • SRTP to protect SIP-associated media

17. SRTP (media) • Defined in RFC 3711 • Confidentiality • Message authentication • Protection to the RTP traffic & control traffic • ZRTP (RFC 6189) • Key exchange protocol used by SRTP • Protection against man-in-the-middle attacks • Better confidentiality & simpler to implement

18. SIPS (signaling) • Defined in SIP RFC 3261 • Send SIP messages over a TLS-encrypted channel • By deploying SIP-based devices that support Secure SIP • Running SIP over TLS on a hop-by-hop basis • RFC 3261 defines the SIPS Uniform Resource Identifier (URI) • Provide secure endpoint-to-endpoint connection • IPsec encrypts data end-to-end: used between SIP entities

19. Availability • European Commission, treatment of VoIP under the EU Regulatory Framework: • Public Available Telephone Service (PATS): • Service available to the public • For originating & receiving national and international calls • Access to emergency services • Interconnexion to the PSTN • Interconnection between VoIP networks via PSTN • Maintenance of network in normal situations: - Integrity and availability of the network - Emergency services - Routing emergency calls

20. Availability in Finland • FICORA regulated • Users able to access the universal emergency call number 112 and the police emergency number 10022 free of charge • Routing of emergency calls to the correct emergency response centre (ERC) • ensuring emergency calls • priority of emergency call • Customers aware of restrictions in telephone service during electricity break

21. Services • Electronic communications services: • Voice • Video • Text chat • Presence • Videoconferencing • Instant messaging (IM): even notifications • SIMPLE: SIP extension to support presence and IM

22. Conclusions • Making data calls much less expensive than voice calls • SIP not a standard, but RFC. SBC needed to manipulate SIP headers in order to normalize SIP protocol • 4G mobile networks 100% IP (VoIP) • IMS needed • Historically, high cost service innovation • Expensive to deploy new services • IMS easier to deploy new services • In the future, network architecture focused on services, not access • IM and presence is one of the killer applications in next generation mobile communications

23. Thanks Questions