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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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slide2
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
what is sip
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
sip default ports and protocols
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
sip terms
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.
why use sip
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.
codecs
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.
dtmf signaling
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
cost savings of sip to pstn
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
why sip t o pstn in my situation
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.
types of sites
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.
other considerations
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
call manager configuration
Call Manager Configuration
  • Media Termination Point
  • Media Resource Group
  • Media Resource Group List
  • Gateway
  • Route Group
  • Route List
  • Route Patterns
config for 2811 running cube
Config for 2811 running CUBE
  • What is CUBE
  • Services and Classes
  • Translation Rules
  • SCCP and dspfarm
  • Dial Peers
  • Sip-ua
what is cube
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
services and classes
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

!

translation rules
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

!

sccp and dspfarm
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

!

sccp and dspfarm cont
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

!

dial peers
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

!

sip ua
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

!

config for 2811 for legacy tdm switch
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
voice service
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

!

t1 c onfig
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

!

dial peers33
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

!

dial peers34
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

!

dial peers35
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

!

sip ua36
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

!

avaya legacy switch config
Avaya Legacy switch config
  • DS1
  • Trunk
  • Signal group
  • Route, ARS, and Cabling
avaya misc
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.
testing
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
troubleshooting resources
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
some issues i had and fix
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
summary
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.