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SIP

SIP. What is SIP (Session Initiation Protocol) Review implementation of SIP to the PSTN (Public telephone network). This includes multiple sites, some using Cisco call Manager with Cisco gateways and other utilizing Cisco gateways to connect to legacy phone systems. Lessons learned.

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SIP

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  1. SIP • What is SIP (Session Initiation Protocol) • Review implementation of SIP to the PSTN (Public telephone network). This includes multiple sites, some using Cisco call Manager with Cisco gateways and other utilizing Cisco gateways to connect to legacy phone systems. • Lessons learned

  2. What Is SIP? • Session Initiation Protocol (SIP) • Is a text based signaling protocol. Developed in 1996 • The SIP protocol is situated at the session layer in the OSI model, and at the application layer in the TCP/IP model. • SIP is designed to be independent of the underlying transport layer; it can run on TCP, UDP, or others. • RFC 3261

  3. SIP Default ports and Protocols • Typically on TCP/UDP port 5060 and/or 5061 • In my case Verizon defined the port per site • All voice/video communications are done over separate session protocols, typically RTP

  4. SIP Terms • SIP User Agents (UAs) are the end-user devices, used to create and manage a SIP session. A SIP UA has two main components, the User Agent Client (UAC) sends messages and answers with SIP responses, the User Agent Server (UAS) responds to SIP requests sent by the peer. SIP UAs may work in point to point mode. Typical implementations of a UA are SIP softphones, SIP hardphones and SIP-enabled ATAs. In my case the SIP UA is a Cisco 2811 acting as a VoIP Gateway. • Proxy, Proxy Server: An intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, which means its job is to ensure that a request is sent to another entity "closer" to the targeted user. Proxies are also useful for enforcing policy (for example, making sure a user is allowed to make a call). A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it. • A registrar is a server that accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles. • A redirect server is a user agent server that generates 3xx responses to requests it receives, directing the client to contact an alternate set of URIs.The redirect server allows SIP Proxy Servers to direct SIP session invitations to external domains.

  5. Why use SIP • Stations – many phone use or support SIP. • Trunking internal – Often used for connection various phone systems or sites. 4 digit dialing, toll avoidance for internal calls. • Trunking to the PSTN – Cost reduction, increased utilization of bandwidth.

  6. Codecs • G729 • 8 kbit/s (32kbit/s with overhead) • G729B includes VAD (silence compression) and comfort noise • Does not support Fax • G711 • 64 kbit/s (84kbit/s with overhead) • Will Support fax (including G3) • I’ve had mixed results with modem use.

  7. DTMF Signaling • DTMF – Dual Tone Multi Frequency (aka Touch Tone) • Codec compression may interfere • h245-alphanumeric – DTMF are sent over the h245 channel as ascii • rtp-nte –DTMF are sent in RTP stream as a named telephony event • sip-notify – DTMF are sent as SIP notify messages

  8. Cost Savings of SIP to PSTN • Cost savings – Reduction of additional TDM gear, Consolidation of Voice and data networks. Free “on-net” calls. • Management Control – Consolidated contracting and invoicing, portal for tracking and management

  9. SIP Allows Consolation of Network

  10. Why SIP to PSTN in My Situation • Cost Savings – $200 to $500 per site per month keeping same number of call paths. Will see increased cost savings with reduced number of lines. Will See increased cost savings with BEST (Burstable Enterprise Shared Trunking) which allows pooling of call paths. • Reroute – Enhanced Survivability. We gained the ability to reroute calls to any other SIP site with no additional cost. • Flexibility – ease of turning up and shutting down services. Ability to assign “local DID” to any location.

  11. Types of Sites • Cisco VoIP sites – 5 sites • Centralized Call Manager it Data center. • Cisco VoIP with 7960, 7940, 7920, 7911 phones. • VG224 analog gateways, VG248 analog gateways. • Primary VoIP GW is 2811 running Cube and acting as SRST • Avaya Definity Sites – 4 sites • Avaya G3s • added 2811 running Cube hand off to T1. • Can consolidate into Wan router, we didn’t. • Why not all sites -We did not implement at all sites due to requirements of ROI, few sites not one our “standard” platform, site too small to justify.

  12. Other Considerations • Network failure – T1 down means no SIP, all eggs in one basket. In our situation normally keep 2 CO and fax for backup in event of T1 failure. • 911 – Works with Verizon SIP. We prefer 911 out CO trunk and fail over to SIP. • FAX – Fax over VoIP can get very tricky. We are doing over g711 supporting G3. • Verizon - T.38 FAX support is planned, but is currently not supported. Group 3 FAX sent over G.711 is supported. The CPE must detect fax based on 2100Hz audio signal, after which it must disable echo cancellation and set the jitter buffer to a static time. • 900/976 – Verizon and many other providers “help” you by blocking 900/976

  13. Call Manager Configuration • Media Termination Point • Media Resource Group • Media Resource Group List • Gateway • Route Group • Route List • Route Patterns

  14. Call Manager Media Termination Point

  15. Media Resource Group

  16. Media Resource Group List

  17. Call Manager Gateway Config

  18. Route Group

  19. Route List

  20. Route Patterns

  21. Config for 2811 running CUBE • What is CUBE • Services and Classes • Translation Rules • SCCP and dspfarm • Dial Peers • Sip-ua

  22. What is CUBE • Cisco Unified Border Element • Formally called Multiservice IP-to-IP Gateway • In software feature sets of IOS look for: IPIPGW • For the 2800 Series: • c2800nm-adventerprisek9_ivs-mz • c2800nm-ipvoice_ivs-mz • C2800nm-ipvoice-mz • c2801-adventerprisek9_ivs-mz • c2801-ipvoice_ivs-mz

  23. Services and Classes voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 redirect ip2ip fax protocol pass-through g711ulaw h323 sip ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! voice class h323 1 h225 timeout tcp establish 3 h225 timeout setup 3 call preserve !

  24. Translation Rules voice translation-rule 1 rule 1 /^90\(1..........\)/ /\1/ rule 2 /^90\(011.*\)/ /\1/ rule 3 /^90\(.......\)/ /1317\1/ rule 5 /^90\([2-9]11\)/ /\1/ ! voice translation-rule 2 rule 1 /^21\(..\)/ /31739521\1/ rule 2 /^317392..../ /3173952181/ rule 3 /^8.../ /3173952181/ ! voice translation-profile SIP translate calling 2 translate called 1 !

  25. SCCP and dspfarm voice-card 0 dspfarm dsp services dspfarm ! sccp local FastEthernet0/0 sccp ccm 10.X.X.X identifier 1 priority 1 version 4.0 sccp ! sccp ccm group 1 description CCM bind interface FastEthernet0/0 associate ccm 1 priority 1 associate profile 2 register SHB-CBRIDGE associate profile 1 register SHB-XCODE associate profile 3 register Shelbyville-mtp ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec pass-through maximum sessions 12 associate application SCCP !

  26. SCCP and dspfarmCont dspfarm profile 2 conference description Conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP ! dspfarm profile 3 mtp codec g729br8 maximum sessions software 24 associate application SCCP !

  27. Dial Peers ! dial-peer voice 100 voip destination-pattern 8... modem relay nse codec g711ulaw gw-controlled voice-class codec 1 voice-class h323 1 session target ipv4:10.10.2.15 incoming called-number 9 dtmf-relay h245-alphanumeric fax rate disable fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco no vad ! ! dial-peer voice 1002 voip description VoIP dial peer to VzB translation-profile outgoing SIP preference 1 destination-pattern 90T session protocol sipv2 session target sip-server dtmf-relay rtp-nte digit-drop codec g711ulaw no vad !

  28. SIP UA sip-ua retry invite 2 retry bye 2 retry cancel 2 retry options 2 sip-server ipv4:172.X.X.X:YYYY g729-annexb override !

  29. Config for 2811 for Legacy TDM switch • Call from/to SIP handed off to/from T1 • Voice service voip • T1 config • Controller • Interface • Voice Port • Dial Peers • Sip-ua

  30. Voice service voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 redirect ip2ip fax protocol pass-through g711ulaw h323 sip rel1xx disable !

  31. T1 Config ! interface Serial0/0/0:23 no ip address encapsulation hdlc no logging event link-status isdn switch-type primary-ni isdn timer T321 30000 isdn protocol-emulate network isdn incoming-voice voice isdn guard-timer 1000 isdn send-alerting no fair-queue no cdp enable ! ! controller T1 0/0/0 framing esf linecode b8zs cablelength short 133 pri-group timeslots 1-24 ! voice-port 0/0/0:23 no non-linear playout-delay maximum 120 playout-delay nominal 15 playout-delay minimum low busyout action shutdown busyout monitor FastEthernet0/0 !

  32. Dial Peers ! dial-peer voice 100 voip description originating voip dial peer translation-profile outgoing SIP preference 5 destination-pattern .T rtp payload-type cisco-codec-fax-ack 114 rtp payload-type cisco-codec-fax-ind 113 rtp payload-type nte 98 session protocol sipv2 session target sip-server dtmf-relay rtp-nte codec g711ulaw ip qos dscp cs5 media ip qos dscp cs3 signaling no vad !

  33. Dial Peers dial-peer voice 200 voip description terminating voip dial peer translation-profile incoming DID_fix rtp payload-type cisco-codec-fax-ack 114 rtp payload-type cisco-codec-fax-ind 113 rtp payload-type nte 98 session protocol sipv2 session target sip-server incoming called-number .T dtmf-relay rtp-nte codec g711ulaw ip qos dscp cs5 media ip qos dscp cs3 signaling no vad !

  34. Dial Peers dial-peer voice 10 pots preference 1 destination-pattern 7... progress_ind alert strip 8 direct-inward-dial port 0/0/0:23 prefix 7 !

  35. Sip-ua sip-ua set sip-status 400 pstn-cause 31 set sip-status 401 pstn-cause 21 set sip-status 403 pstn-cause 21 set sip-status 405 pstn-cause 63 set sip-status 406 pstn-cause 79 set sip-status 410 pstn-cause 22 set sip-status 488 pstn-cause 31 set sip-status 501 pstn-cause 38 set sip-status 503 pstn-cause 41 set sip-status 606 pstn-cause 38 set pstn-cause 6 sip-status 406 set pstn-cause 27 sip-status 502 set pstn-cause 30 sip-status 501 set pstn-cause 31 sip-status 480 set pstn-cause 43 sip-status 502 set pstn-cause 44 sip-status 503 set pstn-cause 49 sip-status 503 set pstn-cause 50 sip-status 503 set pstn-cause 58 sip-status 503 set pstn-cause 63 sip-status 503 set pstn-cause 66 sip-status 480 set pstn-cause 69 sip-status 503 set pstn-cause 70 sip-status 503 set pstn-cause 81 sip-status 502 set pstn-cause 82 sip-status 502 set pstn-cause 83 sip-status 503 set pstn-cause 84 sip-status 503 set pstn-cause 85 sip-status 503 set pstn-cause 86 sip-status 408 set pstn-cause 88 sip-status 503 set pstn-cause 91 sip-status 502 set pstn-cause 95 sip-status 503 set pstn-cause 96 sip-status 409 set pstn-cause 97 sip-status 480 set pstn-cause 98 sip-status 409 set pstn-cause 99 sip-status 480 set pstn-cause 100 sip-status 501 set pstn-cause 101 sip-status 503 set pstn-cause 111 sip-status 500 retry invite 2 retry bye 2 retry cancel 2 sip-server ipv4:172.X.X.X:YYYY g729-annexb override !

  36. Avaya Legacy switch config • DS1 • Trunk • Signal group • Route, ARS, and Cabling

  37. Avaya DS1 config

  38. Avaya Trunk Config

  39. Avaya Signal Group

  40. Avaya Misc • Set up a route pattern to use the trunk group • Set up ARS to use route pattern • Cabling, Had custom cable made for rj45 to Amphenol. No CSU used on Avaya side.

  41. Testing • Make sure to test the following • In bound voice call • In Bound FAX • Outbound • Local (7 digit and 10) • LD • International • 411 or information • Consider testing 911

  42. Troubleshooting resources • http://en.wikipedia.org/wiki/List_of_SIP_request_methods • http://en.wikipedia.org/wiki/List_of_SIP_response_codes • Confirming MTP are up and registered: Show sccp ( to reset no sccp sccp) • sh sip-ua calls • sh voice call status • Debugs • debug ccsip errors • debug ccsip all • debug voip dialpeer all • Number Translations - http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00803f818a.shtml • SIP and PSTN event codes - http://www.cisco.com/en/US/docs/ios/12_2t/12_2t11/feature/guide/ftmap.html#wp1017378 • Verifying and Trouble shooting Sip features - http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/trouble.html

  43. Some Issues I had and fix • One way audio fix was MTP and MRG MRGL. • Call setup but can't answer call Issue fix was Gateway configuration in call manager with Fast start • No ring back from PSTN on some calls outbound (like to cells) fix was Outbound fast start on GW • Calls not completing Issue was Calling from number not in range fix was number translation • One way audio then call drops fix was order of MRG in MRGL • No Ring Back Issue was transcoder had wrong Device pool which effected region • Not passing DTMF fix was dtmf-relay rtp-nte h245-alphanumeric in dial peer (especially in bound) • Intermittently failing SIP calls was related to xcode running out of resources and MRG use resources in round robin. • Call not going out correct GW fix was to reset Route list • ring back issues for CUE Send H225 User Info Message • CUE ring back during transfer of call fix was to set up announcator

  44. Summary • SIP can be a good match • SIP is stable and implementable • Many options for how you implement • In house like Mine • Managed services where GW is managed by provider • Entire PBS is in the providers cloud with just SIP phones • SIP is prime time.

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