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Towards Junking the PBX: Deploying IP Telephony

Towards Junking the PBX: Deploying IP Telephony. Wenyu Jiang, Jonathan Lennox, Henning Schulzrinne and Kundan Singh Columbia University {wenyu,lennox,hgs,kns10}@cs.columbia.edu. We describe our departmental IP telephony installation. Outline. Introduction to IP telephony System architecture

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Towards Junking the PBX: Deploying IP Telephony

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  1. Towards Junking the PBX: Deploying IP Telephony Wenyu Jiang, Jonathan Lennox, Henning Schulzrinne and Kundan Singh Columbia University {wenyu,lennox,hgs,kns10}@cs.columbia.edu We describe our departmental IP telephony installation

  2. Outline • Introduction to IP telephony • System architecture • Call flows • System configuration • Security • Scalability Columbia University, Deploying IP Telephony

  3. Traditional Telecommunication Infrastructure 7040 External line 212-8538080 7041 Corporate/Campus Telephone switch Another switch Private Branch Exchange 7042 7043 Internet Corporate/Campus LAN Columbia University, Deploying IP Telephony

  4. What is IP Telephony ? Corporate/Campus Another campus 7040 8151 External line 8152 7041 PBX PBX 8153 VoIP Gateway 8154 7042 VoIP Gateway 7043 Internet LAN LAN IP Phone Client Columbia University, Deploying IP Telephony

  5. IP Telephony Protocols audio over RTP Call “bob@office.com” SIP server home.com office.com • Contact “office.com” asking for “bob” Session Initiation Protocol - SIP • Locate Bob’s current phone and ring • Bob picks up the ringing phone Real time Transport Protocol - RTP • Send and receive audio packets Columbia University, Deploying IP Telephony

  6. rtspd Quicktime SNMP RTSP media server RTSP sipconf Telephone RTSP clients Web based configuration SIP conference server sipum Telephone switch SIP/RTSP Unified messaging sipd T1/E1 RTP/SIP 3Com SIP proxy, redirect server SQL database Cisco 2600 gateway Cisco 7960 e*phone Hardware Internet (SIP) phones Pingtel sipc SIPH.323 convertor Web server NetMeeting sip323 Software SIP user agents H.323 Architecture Columbia University, Deploying IP Telephony

  7. Web based configuration Call Bob sipd SIP proxy, redirect server SQL database e*phone Hardware Internet (SIP) phones sipc Web server Software SIP user agents Example Call (IP only) • Bob signs up for the service from the web as “bob@cs.columbia.edu” • sipd canonicalizes the destination to sip:bob@cs.columbia.edu • He registers from multiple phones • sipd rings both e*phone and sipc • Alice tries to reach Bob • INVITE sip:Bob.Wilson@cs.columbia.edu • Bob accepts the call from sipc and starts talking cs.columbia.edu Columbia University, Deploying IP Telephony

  8. Canonicalization Bob.Wilson canonicalize bob@cs Columbia University, Deploying IP Telephony

  9. Other Services • Programmable servers • Time-of-day, caller identification • CPL, SIP CGI • Unified messaging • Centralized voice mail • SIP, RTSP • Conferencing • Dial-in bridges; centralized audio mixing • Audio, video and chat Columbia University, Deploying IP Telephony

  10. PBX Gateway PSTN Internal T1/CAS (Ext:7130-7139) External T1/CAS Call 9397134 Call 7134 Ethernet 713x is called a part of Coordinated Dial Plan (CDP) in a Nortel PBX 2 1 3 4 5 Regular phone (internal) SIP server SQL database sipd sipc Bob’s phone 7134 => bob PSTN to IP Call (Direct Inward Dial) • DID - direct and simple • No-DID - dial extension, supports more users Columbia University, Deploying IP Telephony

  11. PBX Gateway (10.0.2.3) PSTN External T1/CAS Internal T1/CAS Call 5551212 Call 85551212 Ethernet 3 4 2 1 5 5551212 Bob calls 5551212 Regular phone (internal, 7054) SIP server SQL database sipd sipc Use sip:85551212@10.0.2.3 IP to PSTN Call Note: In this direction there is no distinction between DID and non-DID calls. Columbia University, Deploying IP Telephony

  12. T1 Line Configuration (From the PBX Side) • Electrical/physical settings • T1 type: Channelized, PRI • Characteristics: line coding - AMI, B8ZS; framing - D4, ESF • Trunk type: DID, TIE • Channel type: Data, Voice-only, Data/Voice • Access permissions: adjust NCOS for internal T1 trunk and CDP routing entry (713x) Columbia University, Deploying IP Telephony

  13. VoIP Configuration in the Gateway: Dial Peers • Dial Peer for PSTN to IP calls: dial-peer voice 1 voip destination-pattern 713. voice-class codec 1 session protocol sipv2 session target ipv4:128.59.19.141 • Dial Peer for IP to PSTN calls: dial-peer voice 1000 pots destination-pattern ((70..)|(71[0-24-9].)) no digit-strip port 1/0:1 • Regular expressions to avoid ambiguity Columbia University, Deploying IP Telephony

  14. Dial Peers for non-DID calls • Example for a mix of DID and non-DID translation-rule 7138 rule 1 71381.% 1 ANY abbreviated dial-peer voice 1 voip destination-pattern 713[0-79] … dial-peer voice 2 voip destination-pattern 7138T translate-outgoing called 7138 … • Caller dial 939-7138, then punch in a 3-digit extension of the form 1xx. Columbia University, Deploying IP Telephony

  15. canonicalize canonicalize using dial plan using dial plan Find tel: uid in SQL Primary User Table Locate user’s contact information Locate proper gateway Verify caller’s privilege VoIP Configuration in sipd:Dial Plan • IP to PSTN call • PSTN to IP call sip:7134@sipd-host sip:5551212@sipd-host tel:+12129397134 tel:+12125551212 sip:85551212@gw sip:bob@sipd-host Columbia University, Deploying IP Telephony

  16. Example Dial Plan • Dial plan mapping for IP to PSTN calls # Intra-department calls 7[01]?? tel:+1212939$ # Local (same area code) calls ??????? tel:+1212$ # Remove dial-out prefix ‘8’ (8)??????? tel:+1212$ # International numbers (011)* tel:+$ (8011)* tel:+$ Columbia University, Deploying IP Telephony

  17. Security • Goal: prevent unauthorized users from making certain (e.g., long-distance) calls • Where to put authentication modules: • In the gateway (requires vendor’s support) • Or, its associated SIP proxy server • Prevent direct calls that bypasses the proxy • Enforce signaling path using IOS access control • SIP authentication • Digest, Basic, PGP Columbia University, Deploying IP Telephony

  18. Gateway Selection and Privileges • Approaches • RFC 2916: ENUM, E.164 based on DNS • RFC 2871: TRIP, allows optimization • Static routing file, used in sipd (+1212939)7[01]?? full,guest sip:$@gw.office.com • full and guest are user’s gateway classes • The server may terminate the call if caller has no sufficient privileges. Columbia University, Deploying IP Telephony

  19. Sample Access Control List (ACL) • Configure NIC to use ACL 101 (in packets) interface FastEthernet0/0 ip address 128.59.19.28 255.255.248.0 ip access-group 101 in • Definition of ACL 101 access-list 101 permit ip host 128.59.19.141 any access-list 101 permit udp 128.59.16.0 0.0.7.255 \ range biff 65535 host 128.59.19.28 neq 5060 • SIP requests (destination port 5060) allowed from only the designated proxy host • Multimedia (RTP) packets treated otherwise Columbia University, Deploying IP Telephony

  20. CINEMA: Columbia InterNet Extensible Multimedia Architecture • Web interface • Administration • User configuration • Unified Messaging • Notify by email • rtsp or http • Portal Mode • 3rd party IpTelSP Columbia University, Deploying IP Telephony

  21. Scalability via DNS SRV • A simple load balancing scheme example.com _sip._udp 0 40 a.example.com 0 40 b.example.com 0 20 c.example.com 1 0 backup.somewhere.com • a and b each receives 40% of total request • c receives remaining 20% • backup server for fault tolerance Columbia University, Deploying IP Telephony

  22. Scalability Continued • 2-stage load balancing based on DNS SRV • Stage 1: stateless routing based on hashing • Stage 2: • Hashed clusters • Stateful proxy • Redirect feature Columbia University, Deploying IP Telephony

  23. Scalability of Media Servers • Media packets => more load than signaling • rtspd: multiple server selection: static/dynamic • sipconf: tree structure • Bandwidth savings similar to multicast • Added packetization and playout delay Columbia University, Deploying IP Telephony

  24. Scalability of Gateway and LAN • 1 T1 line = maximum 24 voice channels • Multiple T1 lines or gateways • IP Centrex service by carrier PBX with ethernet • LAN bandwidth limitations (gateway calls) • Silence Suppression: 40-45% activity factor • Faster Ethernet interface (10 => 100 Mb/s) Columbia University, Deploying IP Telephony

  25. SNMP Support in sipd • sipd status • Details of active transactions • User contact info Columbia University, Deploying IP Telephony

  26. Detailed SNMP MIBs • User contact info • Details of active transactions Columbia University, Deploying IP Telephony

  27. Future Work • Additional services • PIN numbers for telephone users • Automated, electronic billing • Instant messaging • VoiceXML (e.g., email access via PSTN) • Performance and scalability: • sipd, rtspd, sipconf • SQL main-memory vs. disk database • Firewall/NAT interoperability • Details of system to appear in Tech Report Columbia University, Deploying IP Telephony

  28. Conclusion • Initial field test experience with deploying IP telephony in a campus environment • The architecture and installation experience can be used at other organizations • Issues raised for further study: • Service availability/reliability • Quality of Service (QoS) • Privacy/encryption • Electronic billing policies Columbia University, Deploying IP Telephony

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