IP Possibilities Conference & Expo Minneapolis, MN April 11, 2007
Rural VoIP Protocol, Standards and Technologies Presented by: Steven P. Senne, P.E Chief Technology Officer Finley Engineering Company, Inc S.Senne@fecinc.com
Rural VoIP Protocols, Standards and Technologies VoIP is increasing in importance as potential cost savings are identified. This topic will be of interest to professionals involved with planning, implementing and testing rural VoIP networks. The panel will discuss protocols, such as SIP, H.323, and others.
Why VoIP? The Interesting Stuff Telecommunications Act of 1996 Deregulation of the Bell networks Open competitive markets for Service Providers Converged Networks Voice, Video & Data over an IP network Reduced the costs of managing parallel networks Allows voice to be an IP application Centralized or distributed architectures Add features where they are needed Bypasses existing TDM network and cost recovery structures
VoIP? The Challenging Stuff Do we need to replicate all the existing PSTN / PBX features? What s the right architecture? Centralized Distributed Mix of both How do we? Provide better than PSTN QoS Provide Admission Control Secure the signaling & media Meet all the regulatory requirements
Open Packet Telephony TDM/ Circuit Switch Line Concentration Digital Trunk Subsystem Switching Network Call Control Connection Control Features Common Channel Signaling Complex Administration Maintenance Billing Open Service Application Layer (JAIN, AIN, TAPI, JTAPI, XML etc.) Open Call Control Layer (SIP, H.323, MGCP, etc.) Standards-Based Packet Infrastructure Layer (IP, ATM) Open/Standard Interface Open/Standard Interface
VoIP Signaling Protocols H.323 ITU standard, ISDN-based, distributed topology 90%+ of all Service Provider VoIP networks Useful for video applications (i.e.. Netmeeting) MGCP IETF RFC2705 Centralized Call-Control Architecture Call-Agents (MGC) & Gateways (MG)
VoIP Signaling Protocols SIP IETF RFC2543 Session Initiation Protocol Distributed Call-Control Used for more than VoIP Instant Messaging/Presence RTP Real Time Protocol Peer to Peer media Transfer
Basic H.323 Call Gatekeeper A ACF LRQ LCF Gatekeeper B ACF RRQ/RCF IP Network RRQ/RCF ARQ H.225 (Q.931) Setup ARQ H.225 (Q.931) Alert and Connect Phone A V Gateway A H.245 RTP LRQ = Location request LCF = Location confirm ARQ = Admission request ACF = Admission confirm RRQ = Register request RCF = Register confirm V Gateway B Phone B
MGCP Architectures & Mixed Protocols SCP PSTN Gateway SIP or H.323 Network BTS / VSC P S T N SS7 IMT V PSTN SIP H.323 GK PRI V V Access Gateway MGCP RTP SIP / H.323
What is SIP? Session Initiation Protocol (SIP) was defined via RFC2543 on March 17, 1999. Additions made in Sept, 1999 and April 2000 Peer to Peer Communications SIP is a very Internet friendly protocol SIP reuses a lot of Internet protocols & formatting Customers still weary about proprietary protocols Skinny works well, but it is Cisco s proprietary Protocol It s about the Applications!! The next Killer App is the integration of voice, data, video, IM Presence SIP can do this Microsoft intends to support SIP on 250 million desktop PCs SIP client will be added to Windows XP Shareware/Freeware Programs available for Windows and other platform
The various flavors of SIP RFC2543 vanilla SIP The most commonly deployed & developed by commercial vendors SIP-T Inter Call Agent (MGC) protocol for carrying SS7 / ISUP messaging Basically maps ISUP messaging to a MIME attachment SIP extension from PacketCable Additions to Security, QoS & Privacy areas
Basic SIP Call-Flow SIP UA1 SIP UA2 INVITE w/ SDP for Media Negotiation 100 Trying 180/183 Ringing w/ SDP for Media Negotiation MEDIA 200 OK ACK MEDIA BYE 200 OK
Basic SIP Functionality - Call Redirection 392-1234 Where is sip:3921234@cisco.com? Location Database Proxy / Redirect Server You need to contact 4721111 INVITE sip:3921234@cisco.com 3xx Moved Contact: sip:4721111@10.1.1.3 INVITE sip:4721111@10.1.1.3 LOCAL PSTN The user at 392-1234 informed the network that he could be reached on his cell-phone at 472-1111 National PSTN
VoIP Implementation Issues NAT and Firewalls Most currently don t support SIP May be able to originate Calls, but can not receive a call behind the firewall Network Congestion No QOS guarantee over Public IP Networks Calls may not be able to be completed or may be dropped in progress Security Packet CALEA Requirements on Peer to Peer Systems SIP passes usernames unencrypted SIP supports encryption of the call data but is not widely implemented IP Phones can be hacked but security is improving SIP Phones and Gateways can accept Calls without Verifying the Source Spam on Voice Over IP phones!!!
VoIP Implementation Issues Customer Premise Equipment Costs SIP Phones are $150-$350 2 Line Analog Gateways are $100 Software for PC based solutions: Shareware or Freeware Requires High-speed Internet Access ADSL, Cable Modem, Wireless service Can be provided over a dial-up connection but at lower quality 911 Trunking to Local PSAP Calls routed to Local PSAP instead of where the caller is located
VoIP Implementation Issues Power of CPE Life-Line Service Power Over Ethernet Uncertain Regulatory Environment Unknown Cost Recovery
VoIP Security Issues SIP Messages are not encrypted SIP Over Transport Layer Security (TLS) Does not encrypt the actual media Encrypts from Proxy to Proxy, End link may not be protected The TLS protocol(s) allow applications to communicate across a network in a way designed to prevent eavesdropping, tampering, and message forgery. TLS provides endpoint authentication and communications privacy over the Internet using cryptography. Typically, only the server is authenticated (i.e., its identity is ensured) while the client remains unauthenticated; this means that the end user (be that a person, or an application such as a web browser), can be sure with whom they are "talking". The next level of security in which both ends of the "conversation" are sure with whom they are "talking" is known as mutual authentication. Mutual authentication requires public key infrastructure (PKI) deployment to clients. Secure RTP (RFC 3711)
Questions?