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COURSE 240 | 2-DAY SESSION
Hands-On Internetworking with SIP in Converged Networks

Course Outline

Prerequisites for this course are a working knowledge of TCP/IP and an intermediate-level knowledge of VoIP concepts.

Section 1: Introduction & Business Case for SIP

A. How/why SIP is changing the future of IP communications
B. Major players shaping the future of SIP
C. State of the SIP market and development initiatives
D. Current SIP-based deployments: ITSP (pure play, RBOC, IXC, and Cable MSO) and Enterprise
E. The sip.edu initiative of the Internet2 and leading universities
F. How Peer2Peer networking technology (Skype, standards-based P2P, or others) will shape the competitive landscape
G. The possibilities and challenges created by FCC legislation

  1. Impact on ITSPs
  2. Impact on consumers
  3. Where are the revenue opportunities for VoIP, aka“just another IP application?”


Section 2: Core SIP Standards

A. SIP components - UAs, Proxy Servers, Redirect servers
B. SIP (RFC 2543) and SIPbis (RFC 3261) overview
C. SIP URIs - phone numbers, email addresses, single point of contact
D. SIP Methods - INVITE, ACK, OPTIONS, CANCEL, BYE, REGISTER, REFER, MESSAGE,
E. R. PRACK, UPDATE, INFO
F. SIP Registration - RFC 3680, Address-of-record, SUBSCRIBE, NOTIFY and rete-limiting, welcome notices
G. SIP Transport Layer Signaling - UDP vs TCP vs TLS over TCP (SIPS)
H. SIP Call Flows - common call routing permutations, the SIP Trapezoid, RTP streams
I. SIP Message Architecture - inside an RFC 3261 SIP message

  1. To
  2. From
  3. Via
  4. Branch
  5. Max-Forwards
  6. Dialog (Formally Call Leg)
  7. CSeq
  8. Call-ID
  9. Contact
  10. Refer-To, RFC 3515

J. SIP Responses

  1. 1xx Informational
  2. 2xx Final
  3. 3xx Redirection
  4. 4xx Client Error
  5. 5xx Server Error
  6. 6xx Global Failure

K. SIP and SDP, RFC 2327, The Offer/Answer model, RFC 3264
L. Locating SIP servers with DNS - RFC 3263, call routing on an Internet scale, SRV records, NAPTR records
M. SIP load balancing and clustering
N. MIME types, RFC 2046 for SIP

Section 3: SIP Advanced Applications

A. SIP Forking
B. SIP and the firewall, ALGs, Xtunnels, STUN, TURN
C. Combining IM and voice, SIMPLE
D. SIP instant multiparty conferencing
E. Video over IP using SIP
F. SIP Call Processing Language (CPL) for end-user controlled call routing

  1. Verizon Iobi
  2. Vonage
  3. CallVantage
  4. Lingo
  5. Others


Section 4: SIP Security Initiatives

A. Attacks and threat models

  1. Hijacking
  2. DoS
  3. Eavesdropping
  4. Tampering
  5. Impersonating

B. SIP Authentication

  1. User to User
  2. User to Proxy
  3. HHTP Digest Authentication
  4. S/MIME, RFC 2633, Certificates and Key Exchange

C. SIP Header Privacy, RFC 3323
D. TLS, RFC 2246
E. SRTP
F. MIKEY

Section 5: SIP Implementations: Endpoints, Servers, & Software

A. Consumer Hard phone options - Cisco, SNOM, Pingtel, Siemens, Nortel, Mitel, 3com, Radvision, Grandstream
B. Consumer video Hard phone options - Wooksung, 8x8, Viseon
C. Consumer softphone options – Xten Eyebeam SDK, Microsoft initiatives,
D. Consumer gateway options - Motorola, Cisco ATA-186/8, Linksys PAP2, 8x8
E. WiFi enabled SIP phones - Free world dialup, Clipcomm, the Vocera badge, PocketSkype
F. Skype plus Hard phone with Siemens Gigaset USB adapter, Skype/PSTN DualPhone
G. SIP in the enterprise IP PBX market - Cisco, Avaya, Nortel
H. SIP and the Softswitch revolution -
I. Nortel, Lucent, Siemens, Cisco
J. Why the Telcos are replacing their Class4/5 TDM switches
K. Open source SIP servers - parallels to the mainstream adoption of Linux, Asterisk, SIP Express Router

Section 6: SIP Advanced Topics

A. TRIP - Telephony Routing over IP to peer providers
B. Billing and CDRs
C. How MGCP/MEGACO, RFC 3015, and SIP are used together
D. How SIP interoperates with RTSP
E. SIP and CALEA
F. The SIP CGI
G. SIP Resources


HANDS-ON LABS - You must bring your laptop to participate in the labs.

Lab 1: SIP Point-To-Point Calls
Statically configure a SIP endpoint, observe signaling trace via Ethereal, capture the bearer channel to a file and play

Lab 2: SIP Client to Proxy Configuration
Learn the necessary parameters used on SIP endpoints, configure a SIP softphone to register with a SIP proxy, place a call from SIP softphone to SIP gateway

Lab 3: Using the SIP REFER Method
Observe the SIP signaling during a call transfer, trace SIP signal flow using the Record-Route header

Lab 4: SIP Forking Proxy
Follow call routing logic through a call destined for multiple endpoints, observe the effects of authorization procedures on a forked call

Lab 5: Locating SIP Servers
Observe options used to enable call routing on a global scale, configure DNS SRV records for SIP server location, configure DNS NAPTR Records for SIP call setup

Lab 6: SIP Interaction with NAT
Configure endpoints to traverse a symmetric NAT device that is not SIP aware, observe SDP messages that enable NAT traversal





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