Introduction to VoIP

1. Introduction to VoIP Dr. Adeel Akram Telecommunication Engineering Department UET Taxila

2. Course Outline • PSTN • Voice over IP Technology • IP Signaling Protocols • VoIP Applications and Services • Asterisk PBX • Case Study/Term Paper/Research Proposal

3. Text Book • Voice over IP Fundamentals by Jonathan Davidson; James Peters; Manoj Bhatia; Satish Kalidindi; Sudipto Mukherjee • ISBN-10: 1-58705-257-1

4. Course Website • IP Telephony and Voice over IP Course Website • http://web.uettaxila.edu.pk/CMS/SP2010/teVOIPms will publish • Lecture Notes • Quiz Marks • Assignments • Attendance Record • Web Links • Related Software • Email: adeel@uettaxila.edu.pk

5. Introduction to VoIP • Classic Telephony • Data Networks(Review) • VoIP • What it is • Protocols • Hardware • Software • Examples • Web Links

6. Classic Telephony in 1 slide • Classic Telephony • Calls happen by electro-mechanical manipulation of voltage levels between the telco network and end-user phones. • voice “payload” and transport signals must be sent together over the same line/circuit. • “Modern” equipment is electronic, but must work w/older equipment. • This makes upgrades and enhancements difficult and expensive.

7. VoIP in 1 slide • Manipulate bits not volts • everything can now happen inside a PC • Can be all-digital or a mix of digital and analog equipment • Telephone Company not required • Can use low-power, commodity hardware instead of expensive, dedicated gear

8. Introduction to VoIP • Review of TCP/IP • VoIP protocols • H323 • SIP • VoIP hardware

9. TCP/IP Review • 4-layer stack • Packet-switched • Error correcting • Reliable delivery • Designed in 1960s/1970s for use over slow, unreliable analog phone networks • Application layer is the key to VoIP and other commonly used protocols

10. VoIP • Uses formal protocol suites to provide: • Call routing, forwarding, voicemail, etc. • Compatibility w/ legacy PSTN/POTS systems • Quality of Service(QOS) • Done mostly w/ software • Can still do point-to-point calls

11. Benefits of VoIP • It’s all just bits. • VoIP is just another application running over a data network(usually TCP/IP) • Service expansion/integration often just a matter of writing the code, not tearing down a switching center • Per-call costs very low, often free • Equipment is often commodity priced

12. VoIP Concerns • Regulatory issues • Tax revenue issues • Market exploitation and control • Quality & Reliability of Infrastructure

13. VoIP Protocols • H.323 : ITU standard, uses ASN.1 (Abstract Syntax Notation 1) • SIP : IETF RFC 2543, HTTP-like headers • SCCP : “Skinny”: Cisco proprietary protocol • MGCP : Media Gateway Control protocol • IAX : Inter-Asterisk eXchange protocol : Native to Asterisk PBX • Skype: Proprietary protocol based on Kazaa • Others

14. H.323 • Specification defined by ITU • Wide support among all telecom providers & manufacturers all over the world • Industry moving away from using it in new products • Meant to provide gateway for telephony devices into the PSTN

15. SIP • “Session Initiation Protocol” • Designed by Cisco Systems, Inc. • Stand-alone computer-to-computer protocol • Does not presume a PSTN • Calls routed/managed by a SIP Server • No real “official” version, so there are lots of different implementations. • Dominant open VoIP protocol

16. SIP

17. Protocol Details • Most VoIP protocols split signaling from voice data, unlike POTS • Potential firewall issues • QOS/capacity checked before calls are initiated

18. RTP • RTP: Real-time Transport Protocol • RTP is ITU standard H.225.0 and is also IETF RFC 1889 • Used by both H.323 and SIP • Similar approaches used by other protocols • Essentially time-stamped UDP packets • Connects between dynamically negotiated port numbers • RTP and NAT/PAT do not mix well !!!

19. VoIP Hardware • Media Gateways • Telephone adapters • Handsets • Soft-switch/PBX • (Much of this can actually be done on a PC)

20. Media Gateway • The media gateway provides the necessary interface for transporting voice content over the IP network and is the source of the VoIP traffic. • Provides a bridge between VoIP networks and the PSTN • Gateway and Gatekeeper are used interchangeably

21. Media Gateway/Gatekeeper

22. Media Gateway • Media gateway features and services can include some or all of the following: • Trunking gateways that interface between the PSTN network and a VoIP network. Such gateways typically manage a large number of digital circuits. • Cable modem/cable set-top boxes, xDSL devices, broadband wireless devices and other residential gateways that provide a traditional analog interface to a VoIP network. • Small-scale (enterprise) VoIP gateways and other access media gateways that provide a traditional analog or digital PBX interface to a VoIP network.

23. Analog Telephone Adapter • Lets you use regular phone on VoIP network • Avoid dedicated or “locked” adapters if possible • $40-200 US

24. VoIP Phones • (Really an embedded computer with VoIP client software) • Ethernet or wireless (802.11b) • $120-300

25. VoIP Phones • USB Phones(“Skype phones”) • Much simpler, much less expensive ($40-90 US)

26. Softphones • Telephony functions completely in software • Run on desktop, laptop and embedded systems. • Some common softphone clients • SJPhone • Xlite • Kphone • OhPhone

27. VoIP Networks • PSTN Telephony • PC-to-PC • Converged Networks • VoIP-based POTS replacement • VoIP PBX using Asterisk

28. PSTN Telephony Network

29. How Telephone Network Works • PSTN: Public Switched Telephone Network • E.164: ITU standard for “phone numbers” • DTMF: Dual-Tone Multi-Frequency “touch tones”

30. How Telephone Network Works • PBX: Private Branch Exchange • Manages calls into and out of organization • Does phone number translation • ISDN: Integrated Services Digital Network • BRI: Basic Rate, 2 * 64Kbps data channels • PRI: Primary Rate, 2Mbps (E1)

31. How VoIP Works/IP to IP Telephony • All VoIP protocols operate in a similar fashion • Control channel • to set up a call • Media channels • to carry encoded voice data • Similar approach to FTP • Lots of protocols for control and media channels

32. PC-to-PC VoIP Network with IP PBX

33. Converged Network • Converged networking defines a network that has sufficient resources to pass data and voice traffic simultaneously

34. Asterisk • Open Source GPL-Licensed PBX • Runs under most popular versions of UNIX (Linux, BSD, OS X) • Can replace a traditional office PBX • Supports soft phones, SIP handsets, wireless phones, etc.

35. Asterisk • Open Source PBX project, in existence since 1999 • Available on Linux, OS X, Windows • Supports SIP, H.323, H.264, IAX protocols • Can route PC-to-PC, PC-to-PSTN, PSTN-to-PSTN calls • Scriptable/Programmable

36. Asterisk Applications • Actions are really applications/programs • Dial(), Playback(),Voicemail() • Custom applications can be written using AGI, “Asterisk Gateway Interface” • Support for Perl, Python, C, etc. • Uses Linux stdio to pass data between external applications and Asterisk

37. Voice CODECs • A CODEC(enCOder/DECoder) is a method or algorithm for processing analog audio signals in a digital data stream • Most commonly used for processing audio data sent over a network but can be streamed from a file

38. Common VoIP CODECs

39. Configuration Issues on IP Networks • SIP-registration/setup on port 5060/5061 • H.323 defaults to ports 1718/1719 • IAX registration defaults to 4559 • Typically runs into problems with multiple NAT layers

40. PC to PC VoIP Network • How to locate the other phone? • Static configuration is possible — but doesn’t scale • In H.323 a directory server is commonly used • In SIP a proxy server can provide directory services via redirection

41. SIP Information Flow

42. SIP Proxy • Another common SIP proxy approach • Proxy in the middle of all control communication • Note how media channels still flow directly

43. VoIP and Firewalls • VoIP control channel is usually a single well known port • H.323: TCP and UDP 1720 • SIP: TCP and UDP 5060 • Other ports can be used as the port number included in the protocol addresses • Media channels are dynamically negotiated, often within a wide range of ports • Assumes the “end to end” Internet • Can lead to “one way audio”

44. The Challenge of NAT/PAT • Control channel can usually be NAT’d through firewall okay • But media channel is challenging • Because dynamic port negotiation includes IP addresses • Meaningless outside the LAN if using RFC 1918 addresses (Private) • Typical symptom is “one way audio” • If both ends have the problem then no audio will be heard • This is a moderately common issue with FTP as well, but there is better firewall support for FTP

45. NAT/PAT solutions • Using a protocol aware firewall • For Linux, sip-conntrack-nat: http://www.iptel.org/sipalg/(Alpha test code; in iptables Patch-o-Matic) • For Linux, h323-conntrack-nat:http://max.kellermann.name/projects/netfilter/h323.html(Alpha test code; in iptables Patch-o-Matic) • Using an application level media proxy • For Linux, Asterisk: http://www.asterisk.org/ • Or siproxd: http://siproxd.sourceforge.net/ • Or for H.323: OpenH323Proxy:http://openh323proxy.sourceforge.net/ • Or for H.323: GnuGK or OpenH323GK: http://www.gnugk.org

46. NAT/PAT Solutions • STUN: IETF RFC 3489: Simple Traversal of UDP through NAT • Tunneling in unfiltered, globally unique, IP address • Using vtun or another VPN • Will need to do policy routing to send traffic from those IP addresses and back out of the tunnel • Linux: use iproute2 to route based on the source address range(http://lartc.org/howto/lartc.rpdb.html)

47. Web Links: Open Source VoIP Projects • Asterisk(SIP-based PBX) • www.asterisk.org • AstLinux • www.astlinux.org • OpenH323(H.323) • www.openh323.org • SJ Phone • www.sjlabs.com • Xlite • www.counterpath.com • Mesh Network Hardware • www.belairnetworks.com

48. Questions • ????