brkucc-2931 -...

© 2012 Cisco and/or its affiliates. All rights reserved.BRKUCC-2931 Cisco Public

Centralized SIP Trunk Best PracticesBRKUCC-2931

2


Housekeeping Notes

Please silence your cell phones

This is a non smoking Venue

Use the recycle bins where appropriate

Please Visit the World of Solutions

Questions?

3


Session Objectives

Understand Centralized SIP architecture

Learn about alternative Dial Plan solution

(pros and cons)

Show design challenges for a centralized architecture

Identify possible issues of Centralized SIP Trunk

Share design ideas and lessons learned

4


Scope and Context

Design and lessons learned can be applied to

Enterprise environment

It is not about SBC boxes or SIP Service

Provider architecture

First Enterprise design was done in Nov/Dec

2008

Showing a simplified version of the designs

(several years in 2 hours)

5


Agenda

Typical Deployment Models

Design Challenges

Dial Plan Design

Call Routing Design

Implementation Issues

6

Customer Requirements


Centralized call control captures an additional 4-8% TCO savings

Cuts administrative overhead by centralizing management and minimizing resources required to

update/change services

SIP Trunking provides additional TCO savings of up to 12-26%

• Lowers costs by reducing the number of PSTN circuits

• Reduce administration & maintenance costs of supporting multi-vendor PBXs

• More flexibility & less complexity when provisioning for traffic patterns

Lower Total Cost of Ownership

8

Comparison of Monthly Voice Connectivity Operating Costs Per Seat (TDM versus SIP)

Cost Category TDM With SIP Trunking

DID $4-6 $5-8

Channel Cost PRI $27-37$3-6.8 (assumes a 5x to 8x reduction in channels

based on concurrent call density)

MPLS data service Not applicable $1.4-2.8 (assumes each branch has 10 to 20 users)

SIP Equipment Not applicable $1-2.1

Total Cost $31-43 $10.4-19.7


Estimate Your Own Savings Potential From

SIP Trunking: Use the ROI Calculator

Check out: www.cisco.com/go/cube under “SIP Trunking Cost Reduction Tool”More information can be found at: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640white_paper_c11-633868.html

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http://www.cisco.com/go/cube


Typical Customer Profile

Multiple Site Locations

Centralized IT Management Structure

PBX / Key system Installation base

Existing Converged Networks

High Trunking Cost

11


Deployment Scenarios

Centralized Deployment Model

‒ Central SIP trunk (Centralized Signaling and Media)

Distributed Deployment Model

‒ SIP trunks per location (Direct Signaling and Media)

MPLS Deployment Model

SIP trunks per location (Direct Signaling and Media)

Hybrid Deployment Model

‒ Centralized, distributed, PSTN

12


PSTN

GK

MPLS

SBC SBC

West East

CoW C1

CoW C2

Cluster C1

A

Cluster C2

A

SBCSBC

GK

PSTN

PSTN

Topics Not on the Scope

13


GK

MPLS

PSTNWest

SBC

East

Cluster 1

A

SBC

GK

Datacenter E

Cluster 2Cluster 2

ASBC

SBC

Cluster 1

Traditional Topology

Flow Throughor

Media Anchored

14


PSTNPSTN

GK

MPLS

SBC SBC

West East

SBCSBC

GK

X

PSTN

RTP

CoW C2

CoW C1

Cluster C2

A

Cluster C1

A

Voice Path (RTP)

Flow Aroundor

Media Release

15


PSTNPSTN

GK

MPLS

SBC SBC

Datacenter W

SBCSBC

Datacenter E

GK

PSTN

West East

CoW C2

CoW C1

Cluster C2

A

Cluster C1

A

Signaling Path

17


GK

MPLS

PSTNWest

SBC

East

Cluster 1

A

SBC

Cluster 1

SBC

Datacenter E

GKCluster 2Cluster 2

ASBC

Deployment Scenarios - Centralized

Flow Throughor

Media Anchored

Datacenter W

Signaling

18


GK

MPLS

IP PSTN

Cluster 1

A

Cluster 1

Datacenter E

GKCluster 2

Cluster 2

A

Deployment Scenarios – Distributed

Datacenter W

SBCSBC

PSTNSBC

RTP

• SRST• MTP

• CUCME• SRST• MTP

RTP

RTP

20


PSTNPSTN

GK

MPLS

SBC SBC

West East

SBCSBC

GK

PSTN

CoW C2

CoW C1

Cluster C2

A

Cluster C1

A

Deployed Scenarios - Hybrid

Flow Aroundor

Media Release

Flow Throughor

Media Anchored

SBC IP PSTN

21

Design Challenges


Centralized SIP Trunk Design Challenges

24

DTMF

SRST

Non Ported DIDs

Central Site Device Pool

MoH

FAX


MoH


25


PSTNPSTN

GK

MPLS

SBC SBC

West EastDatacenter W

SBCSBC

Datacenter E

GK

PSTN

Hold

CoW C2

CoW C1

Cluster C2

A

Cluster C1

A MoHMoH

MoH

Max Hop (TTL) = 1

Multicast Addr: 239.1.1.1

RTP Port: 16348

Multicast Addr: 239.1.1.1

RTP Port: 16348

X

MoH with Local PSTN Access

26


PSTNPSTN

GK

MPLS

SBC SBC


SBCSBC

Datacenter E

GK

CoW C2

CoW C1

Cluster C2

A

Cluster C1

A MoHMoH

RTP

-M

oH

Flood Network with Multicast MoH Traffic

from All Sites

Multicast Between Customer and Service

Provider

One Single PSTN Trunk

X

MoH for Centralized SIP Trunk (Distributed)

27


IP PSTN

GK

MPLS

SBC SBC


SBCSBC

Datacenter E

GK

PSTN

Hold

CoW C2

CoW C1

Cluster C2

A

Cluster C1

A MoHMoH

MoH for Centralized SIP Trunk (Unicast)

28



29

Central Site Device Pool

MoH Centralized MoH Configurable to 250 streams


MPLS

PSTN

Cluster C1

A

Unity

GW

Site A

PSTN

HQ

MoH

Device Pool and Region (CODEC)

HQ Device Pool

‒ GW-HQ or SIP trunk, Unity VM, IP

phones, MoH server

‒ G.711/G722 internally, G.729 with other

regions

Site A Device Pool

‒ GW-Site-A, IP phones


regions

30


MPLS Cluster C1

A MoH

Site A

DC

Unity

Cluster 2

A

XX


DC Device Pool

‒ Unity VM, IP phones, MoH

server, SIP trunk

‒ G.711/G722 internally, G.729 with

other regions

31

PSTNI P PSTN


MPLSCluster C1

A MoH

Site A

DC

Unity

Cluster 2

A


DC Device Pool

‒ DC IP phones only

‒ G.711/G.722 internally, G.729 with other

regions

Multiple Device Pools

‒ Unity, SIP trunks, ICT trunks, MoH server

‒ G.729 only

Site A Device Pool

‒ GW-Site-A, IP phones


regions

32

PSTNI P PSTN


Media Resources

Central Site MRGL

‒ SIP Application Integration

‒ Greater Flexibility of Media

Resources

‒ More Segmented Control

‒ MRGL Per Sip Trunk

33

Unity Contact Center Meeting Place Microsoft OCS VCS

I P PSTN

Cluster 2

A

SME

A

MPLS LANSBC

33



34

Non Ported DIDs

Central Site Device PoolMultiple Device Pools for Devices on Datacenters



SIP trunk provider will not own the DID for some sites

Incoming calls require a local TDM trunk with local carrier

Outbound calls (local, 911 and TFN) cannot be routed via SIP trunk

Consequence

Customer still wants to send outbound calls via SIP trunks

Requirement

Depending on the region, some DID cannot be ported to the SIP Trunk

CLEC does not have an agreement in place to port numbers away with the current ILEC or CLEC providing the service

Restriction

DID Not Ported to the SIP Trunk

35


IP PSTN

GK

MPLS

SBC SBC

West East

SBCSBC

GK

CoW C2

CoW C1

Cluster C2

A

Cluster C1

A

Call Flow (DID Ported)

36

Inbound SignalingOutbound SignalingOutbound Signaling


IP PSTN

GK

MPLS

SBC SBC

West East

SBCSBC

GK

TDM

RTP

Inbound Signaling

PSTN

CoW C2

CoW C1

Cluster C2

A

Cluster C1

A

Call Flow Inbound (To DID Not Ported)

37


IP PSTN

GK

MPLS

SBC SBC

West East

SBCSBC

GK

Outbound Signaling

X

LD Trunk (Outbound)• LD• International

PSTNPSTN

CoW C2

CoW C1

Cluster C2

A

Cluster C1

A

Call Flow Outbound (From DID Not Ported)

38


IP PSTN

GK

MPLS

SBC SBC

West East

SBCSBC

GK

TDM

RTP

• Local Calls• 911 • TFN

Outbound Signaling

PSTN

CoW C2

CoW C1

Cluster C2

A

Cluster C1

A

Call Flow Outbound (From DID Not Ported)

39



40

Requires a Different Call Flow and Different Call Routing

FAX

Non Ported DIDs




FAX over SIP

42

Centralized fax server with G.711 to SIP trunk

Need dedicated SIP trunks for Fax DID forcing G.711

Distributed fax with G.711 between SIP trunk and remote sites

Need to reserve BW for Fax (CAC + QoS)

Fax can run over the same SIP trunk as voice

Need to detect fax tone and change codec with re-invite

Non centralized FAX still needs BW reservation for G.711

Centralized/Distributed T.38 Fax

Carrier support Varies


MPLS

IP PSTN

Cluster C1

AMoH

Site A

DC

Unity

Cluster 2

A

N x FXO

(or) PRIPSTN

FAX

Fax Call Flow (FXO/PRI)

43


Support for FAX T.38

44

CUCM

A I P PSTNSBC

FAX

MGCPSIP SIP

MGCP

no ccm-manager fax protocol cisco no mgcp fax t38 inhibit mgcp package-capability fxr-packagemgcp default-package fxr-package

CUBE

voice service voipfax protocol t38 ls-redundancy 0 hs-

redundancy 0 fallback none voice service voip allow-connections sip early-offer forced

Delay OfferDelay to Early

Offer Conversion

FAX T.38



45

SRST

Based on Carrier Support


FAX

Non Ported DIDs




4G WAN Cluster C1

A MoH

Site A

DC

Unity

Cluster 2

A

N x FXO

PSTN

FAXSRST

911 Call BackMain Number

FXO Port 911 Only

Call Flow on SRST (Small Site)

46

IP PSTN


4G

Call Flow on SRST (Medium/Large Site)

47

WAN Cluster C1

A MoH

Site A

DC

Unity

Cluster 2

A

N x PRI

PSTN

FAXSRST

Inbound Main Number

911Local/LDIntl/TFN

IP PSTN

MAN

• With MAN and WAN redundancy –Networking Planning becomes Vital

• Need for Symmetrical Links between MAN and WAN



48

DTMF

Limited Access via FXO (Small)

PRI for Medium/Large Site

SRST

Not Supported on SIP Trunk;

Handled by Site GW


FAX

Non Ported DIDs

Central Site Device Pool Multiple Device Pools for Devices on Datacenters

MoH Centralized MoH Limited to 50


DTMF over SIP Trunk

49

H.323 SCCP

RFC2833XIP PSTN

Cluster 1

A

SBCSBC

SIP

SIP SCCP

RFC2833

IP PSTN

Cluster 1

A

SBCSBC

SIP

MTPX

Check Phone Models

Check Phone Firmware

CUCM Only Accepts RFC2833 If There Is a

SIP Call Leg

It Was Originally Designed with H.323



50

SIP Trunk and Check IP PhoneDTMF

Limited Access via FXO (Small)

PRI for Medium/Large SiteSRST

Not Supported on SIP Trunk;

Handled by Site GW


FAX

Non Ported DIDs


MoH Centralized MoH Limited to 50

Dial Plan Design


• 10 Digit DN, PSTN Like • 9 as access code for PSTN calls

DN Cannot Start with 9

• 8 + 10 digit DN

User Doesn’t Know If Number Is on PSTN or Another Branch

User Doesn’t Know If Remote Branch Has Been Migrated• 9 + 10 digit DN

• 9 as access code for PSTN calls• 4 Digits for same site dialing (last 4 of DN)

4-Digit Extension Cannot Start with 9

• Same site can use:* + last 4 digits of DN

User Will Always Dial 9

• Can use 0 for operator• Can use leading 1 for internal/service numbers

Dial Plan Design

53


Dial Plan Restriction

10-digits dialing for local and long distance calls

Eleven digit and seven digit dialing will not be accepted

All IP Phones will have a 10 digit directory number (DID)

Users will dial 10-digits for IP phones on any cluster

Internal calls to any site will be routed directly via WAN

Same site calls will use an access code plus 4 digit extension

Reserve a range of non-DID DN for future CC agents

Design Two Dial Plans and Present Pros/Cons

Customer Request

54


Dial Plan Option 1 (Access Code)

Option 1 (DN=10 digit) Access Code 9 and *

First Digit User Dial Description Type

0 0 Operator TP

1 100XXXXXXX CTI Ports/RP… DN/RP

1 1[1-3]XXX Call Park DN

1 14XXXXXXXX Contact Center Extensions DN

1 15XXXXXXXX Non DID DN DN

9 9.911 Emergency RP (UP)

9 911 Emergency RP (UP)

9 9.[2-9]XXXXXXXXX LD and DN RP/TP

9 9.011! / 9.011!# International RP

# #XXXXXXXXXX Forward VM DN

* *XXXX Site Extension TP

2-8 Not Used Free

55


Dial Plan 1 (Access Code)

Two Types of Access Code:

• “9” for PSTN calls and Internal DN on other sites/cluster

• “*” for same site extension dialing

Limitations:

• Corporate directory: User must edit to dial

• Dial back (missed call and received calls): Use must edit to dial

• Discard 9 before routing internal 10-Digit DN: Requires a TP

• System cannot distinguish between PSTN and DN

• Existent 911 misdialing issues

• Cannot use “*” for analog phones or International (+)

56


Dial Plan Option 2 (PSTN Flat)

Option 2 (DN=10 Digit) PSTN Flat

First Digit User Dial Description Type

0 00 Operator TP

0 011! / 011!# International RP

1 100XXXXXXX CTI Ports/Unity Ports/RP… DN/RP

1 1[1-3]XXX Call Park DN

1 14XXXXXXXX Contact Center Extensions DN

1 15XXXXXXXX Non DID DN DN

9 911 Emergency RP (UP)

# #XXXXXXXXXX Forward VM DN

* *XXXX Site Extension TP

[2-9] [2-9]XXXXXXXXXX LD and DN DN/TP

57


Dial Plan 2 (PSTN Flat)

User Dialing:

• Plain 10 digit for PSTN calls and Internal DN on other site/cluster

• “*” for same site extension dialing

Limitations:

• System cannot distinguish between PSTN and DN

• Cannot use “*” for analog phones or International (+)

• Cannot reserve “0” for Operator

• Different user experience: No Access Code, no Sec Dial Tone

58

Call Routing Design


Regional Cluster Architecture

61

PSTN

SBC 2

Cluster C4

A

Cluster C1

A

Cluster C3

A

Cluster C2

A

SBCSBCSBC

SBC

Cluster C5

A

SBC 1

VA: 276, 434, 540, 571,703, 757, 804

WV: 304

GA: 229, 404, 470, 478, 678,706, 762, 770, 912

AL: 205, 251, 256, 334SC: 803, 843, 864

NC: 252, 336, 704, 828, 910, 919, 980

540-222-5555 919-444-2222

404-555-6666

Less than 200 Patterns

Less than 200 Patterns

SME

A


CSS-SITE-X-LD(Device)

Site Specific

PT-Site-X

Non DID DN *222X

Call Park 11XX

OperatorTP 00 => 10 Digit DN

Site ExtensionTP *SSXX => 10 Digit DN

CER

RL-ENTIPCCE (Reserved)RP 14XXXXXXXX

BlockRP [2-8]11

RP [79]00XXXXXXXRP 976XXXXXXX

RP [2-9]XX976XXXX

TP RP 911

PT-Global

RGSBC

2

SBCEAST

SBCWEST

PSTN SBCEAST

PSTN SBCWEST

PT-Internal

All Cluster DNDID [2-9]XX-XXX-XXXX

Non DID 100-XXX-XXXXUCCE 14XXXXXXXX

Unassigned DNTP [2-9]SS-SSS-SSXX

=> 100999????Unity Call Handler

IP PhonesRoute Points

RL-GK RG-GK1

Cluster C4

Cluster C5

Cluster C6

Cluster C7GK-West

GK-East

Cluster C3

Cluster C2

PT-LD

RP [2-9]XX-XXX-XXXX

Cluster Area CodesRP AAA-XXX-XXXX

InternationalRP 011! / RP 011!#

TFN: RP 8FF-XXXXXXX

RLSBC

Outbound Call

63


919-555-4444

Site Specific


PT-Internal



Unassigned DNTP 919-555-44XX=> 1009991000

Unity Call Handler

PT-Site-X

Non DID DN *222X

Call Park 11XX


Site ExtensionTP *44XX => 91955544XX

PT-LD

RP [2-9]XX-XXX-XXXX

Cluster Area CodesRP 919-XXX-XXXX


IPCCE (Reserved)RP 14XXXXXXXX

BlockRP [2-8]11


RP [2-9]XX976XXXX

TP RP 911 CER

RL-GK

RL-ENT

RG-GK

RGSBC

1

2

RLSBC

PT-Global

SBCEAST

SBCWEST

PSTN SBC EAST

PSTN SBCWEST


Cluster C4

Cluster C5

Cluster C6

Cluster C7

Cluster C3

Cluster C2

TFN: RP 8FF-XXXXXXX

919-555-4444

408-555-4444

919-777-6666

Outbound Call (Example Same Cluster)

64

GK-West

GK-East


919-555-4444

Site Specific


PT-Internal



Unassigned DNTP 919-555-44XX=> 1009991000

Unity Call Handler

PT-Site-X

Non DID DN *222X

Call Park 11XX


Site ExtensionTP *44XX => 91955544XX

PT-LD

RP [2-9]XX-XXX-XXXX

Cluster Area CodesRP 919-XXX-XXXX



BlockRP [2-8]11


RP [2-9]XX976XXXX

TP RP 911 CER

RL-GK

RL-ENT

RG-GK

RGSBC

1

2

RLSBC

PT-Global

SBCEAST

SBCWEST

PSTN SBC EAST

PSTN SBCWEST


Cluster C4

Cluster C5

Cluster C6

Cluster C7

GK-East

Cluster C3

Cluster C2

TFN: RP 8FF-XXXXXXX

408-555-6666

408-555-6666

408-777-8888

Outbound Call (Example Different Cluster)

65

GK-West

GK-East


Site Specific


PT-Internal





PT-Site-X

Non DID DN *222X

Call Park 11XX



PT-LD

RP [2-9]XX-XXX-XXXX




BlockRP [2-8]11


RP [2-9]XX976XXXX

TP RP 911 CER

RL-GK

RL-ENT

RG-GK

RGSBC

1

2

RLSBC

PT-Global

SBCEAST

SBCWEST

PSTN SBC EAST

PSTN SBCWEST


CSS-SITE-X-ENT(Device)

Cluster C4

Cluster C5

Cluster C6

Cluster C7GK-West

GK-East

Cluster C3

Cluster C2

TFN: RP 8FF-XXXXXXX

Outbound Call (Additional Requirements)

66

GK-West

GK-East

PT-ENT RP [2-9]XX-XXX-XXXX


Site Specific


PT-Internal





PT-Site-X

Non DID DN *222X

Call Park 11XX



PT-LD

RP [2-9]XX-XXX-XXXX




BlockRP [2-8]11


RP [2-9]XX976XXXX

TP RP 911 CER

RL-SME RG-SME

1

PT-Global

SBCEAST

SBCWEST

PSTN SBC EAST

PSTN SBCWEST


CSS-SITE-X-ENT(Device)

PT-ENT RP [2-9]XX-XXX-XXXX

Cluster C4

Cluster C5

Cluster C6

Cluster C7

GK-West

GK-East

Cluster C3

Cluster C2

TFN: RP 8FF-XXXXXXX

Outbound Call (SME)

67

SME

A


Cluster C4

Cluster C5

Cluster C6

Cluster C7

Cluster C3

Cluster C2

PT-Internal

CSS-INCOMING

PT-911-CallBack


Non DID 100-XXX-XXXXUCCE 14XXXXX


=> 100999????

TP <ELIN>913XXXXXXXXXX(One TP per Site)

SBC

SBC

IP PSTN

PSTN GW

Incoming Calls (Cluster)

68

GK-West

GK-East

Implementation Issues


SIP Trunk Lessons Learned

70

Voice Quality on Outbound Call Leg

Voice Quality of Recorded VM Messages/Greetings

Outbound Calls Rejected

No Outbound Calls to SBC

Incoming Calls from SBC Stopped Working


No Outbound Calls to SBC

71


SIP Trunk Issues During Pilot

77

No Outbound Calls to SBC Check If SBC Uses TCP or UDP



SIP Trunk for Incoming PSTN Calls

79



80



Which SIP Trunk Handles

Incoming Calls? Check CSS



MPLS

IP PSTN

Cluster 1

A

Site A

DC

Unity

Cluster 2

A

SBC

SBC

PSTN User Incoming audio to IP Phone is fine but

PSTN user receives bad audio

Problem is intermittent

Inbound and Outbound calls

Can happen to any cluster

Issue

Outbound RTP not always correctly marked

Switch configured to mark all RTP packets based on RTP port number (range 16384K to 32768K)

Troubleshooting

Phone sends RTP on non standard port range as requested by far end (SBC)

SBC configured to use port range 10,000 to 40,000

Cause

SBC Chooses Listening RTP Port

IP Phone Sends Audio to SBC Selected RTP

Port

Packets Marked as BE May Be Dropped

RTP May Be Remarked as BE


81



82

Voice Quality on Outbound Call Leg QoS => Check UDP Port Range







MPLS

IP PSTN

Cluster 1

A MoH

Site A

DC

Unity

Cluster 2

A

Message or Greeting recorded had lower quality.

Issue

PSTN calls and IP phone calls are G.729.

CUC configured to record messages with G.729

Incoming G.729 audio on Unity had good quality.

Outbound G.729 audio had low quality.

Troubleshooting

CUC converts all audio to PCM before recording on selected format.

Changing CUC to record greetings and messages with G.711 fixed the problem.

Cause

Quality of Voice Mail Messages

83



84



QoS => Check UDP Port Range

Record Messages with G.711






No DTMF on PSTN Cross Cluster Calls (RFC2833)

No Ring Back on Blind Transfer

Other SIP Trunk Customer on Same Provider May (VoIP) Require G.711

Reroute on Reject Broken

Media Cut Through Delay

No Ring Back on Outbound Calls

SIP Trunk Issues During Pilot (Part 2)

85

“Very Early Media”


SCCP

Cross Cluster DTMF

86

SCC

P

X

X

SIP

RFC2833H.323/SCCP

SIP

RFC2833

IP PSTN

Cluster 1

A

SBCSBC

SIP

Unity 2

H.323

Cluster 2

A

Unity 1

SIP

RFC2833

Cluster 1

AUnity

H.323

Cluster 2

A

SCC

P

SCCP


Cross Cluster DTMF (Solution)

87

RFC2833RFC2833

SIP Unity Pilot

SIP SCCP

RFC2833

IP PSTN

Cluster 1

A

SBCSBC

SIP

Unity 2

H.323

Cluster 2

A

Unity 1

SCC

P



Dedicated SIP ICT Trunk to Route Unity RP Across Clusters




90


SP

VoIP

Delayed Offer to Early Offer Interworking

91

INVITE (Offer SDP)INVITE

180/200 (Answer SDP)180/200 (Offer SDP)

ACK (Answer SDP)

SBC

SBC

Session Established Session Established

Requires Media Flow Through (Media

Anchored)

SBC Sends Its Own IP Address and RTP Port

on SDP


SP

VoIP

Delayed Offer to Early Offer (Flow Around)

92


180 (Sess. Progress SDP)180 (Offer SDP)

ACK (Answer SDP)

SBC

SBC

SBC Sends Its Own IP Address and RTP Port

on SDP

Session Established

SBC Sends SDP with IP Address/Port of IP

Phone

200 OK

200 OK

Re-INVITE (New SDP)

RTP (Early Media)


Flow Around(Media Release)

IVR

CUCM Send ACK with SDP


SP

VoIP

Delayed Offer to Early Offer (PRACK)

93


180 (Sess. Progress SDP)183 (Offer SDP, 100rel)

PRACK (SDP)

SBC

SBC

Session Established

200 OK

200 OK Re-INVITE (New SDP)

RTP (Early Media)

IP Phone Opens Send and Receive

Channel

IVR






Configure UCM to Use PRACK

(SIP Rel1XX Enabled)


94


Fast Answer


SP

VoIP

Delayed Offer to Early Offer (Very Early Media)

95


180 (Sess. Progress SDP)183 (Offer SDP, 100rel)

PRACK (SDP)

SBC

SBC

Session Established

200 OK

200 OK

Re-INVITE (New SDP)

IVR

RTP (Very Early Media)


SP

VoIP

CUCM 8.5 Early Offer

96

INVITE (Offer SDP)

180 (Sess. Progress SDP)183 (Offer SDP, 100rel) SBC

SBC

Session Established

200 OK200 OK

IVR

RTP (Very Early Media)



No Ring Back on Blind Transfer


Annunciator

Reroute on Reject Broken Caused by Annunciator

Need Dedicated MRGL



Configure UCM to Use PRACK

(SIP Rel1XX Enabled)


97

Flow Through

Media Anchored


Fast Answer

Sip Normalization


Diversion Header Manipulation - CER

99

Called Number 911 Called Number 911

IP PSTN

Cluster 1

A

SBCSBC

CER

CER Intercepts 911 Call and

Adds Diversion Header

Diversion Header 912


Diversion Header Manipulation - CER

100

Called Number 911Called Number 911

IP PSTN

Cluster 1

A

SBCSBC

CERScript

voice class sip-profiles 2request INVITE sip-header Diversion remove!dial-peer voice 1911 voipdescription ** OUTBOUND To VZB PSTN Emergency **destination-pattern 911session protocol sipv2session target sip-servervoice-class codec 1voice-class sip profiles 2dtmf-relay rtp-nteno vad

SIP Profile for Header

Manipulation


CLIN:9195001234 CLIN:9193925731CLIN:9193925731

Diversion Header Manipulation – Non Port DID

101

IP PSTN

Cluster 1

A

SBCSBC

Site Has Numbers that have not been ported

Inbound Calls Still come in Via PSTN

800 Calls were being routed incorrectly

Billing Centers

Issue

PSTN

M = {}function M.outbound_INVITE(msg)local pai = msg:getHeader("P-Asserted-Identity")if pai

thenlocal newpai = string.gsub(pai, ":%d+@", ”:9195001234@")local div =

(newpai..";reason=unconditional;privacy=off;screen=yes")msg:addHeader("Diversion", div)

endendreturn M

Solution

Not Recognized


To:[email protected]:[email protected]

Invite Manipulation - VCS

102

Cluster 1

A

[email protected]

VCS not under control of Telecom

VCS Expects FQDN in invite

VCS was rejecting inbound Video Calls

Need to normalize the SIP invite into VCS

Issue

8.6 and beyond can send FQDN in SIP invite

M = {}function M.outbound_INVITE(msg)trace.enable()local method, oldHeader, ver = msg:getRequestLine()if oldHeader

thenlocal newHeader, ctr = string.gsub(oldHeader, "10.26.244.197:5060",”cisco.com")

msg:setRequestUri(newHeader)msg:modifyHeader("To", newHeader)

endendreturn M

Script

1234

To:[email protected]

102

http://medcity.net/

http://medcity.net/

http://medcity.net/

http://medcity.net/


SDP Pass-through

103

IP PSTN

Cluster 1

A

SBC

WAN

CVP

G.711 VXML

G.729 Agent

Script

VVVV

G.711

G.729

G.729

dial-peer voice 2000 voipdescription ** From PSTN DID and TFN INBOUND **translation-profile incoming NORMALIZE-CALLINGsession protocol sipv2session target sip-serverincoming called-number [2-9]..[2-9]......$voice-class codec 1voice-class sip pass-thru content sdpdtmf-relay rtp-nteno vad

103


Re-INVITE Ptime:20ms

Calling Setup – Packet Rates

104

SP

VoIPSBC

SBCRe-INVITE Ptime:30ms Re-INVITE Ptime:30ms

Session Established Ptime:20ms

SBC SP

VoIPSBC


Session Established Ptime:20ms


ScriptM = {}function M.outbound_INVITE(msg)

local sdp = msg:getSdp()if sdpthen

-- Normalize all a=ptime lines with a=ptime:20 per IPT requirementsdp = sdp:gsub("a=ptime:%d*\r\n", "a=ptime:20\r\n")-- store the updated sdp in the message objectmsg:setSdp(sdp)

endendreturn M

104

Final Design Thoughts


PSTNSBC 2 SBC 3

Cluster C4

A

Cluster C1

A

Cluster C3

A

Cluster C2

A

SBCSBC

SBCSBC

Testing New Features

107

Cluster C5

A

SBC 1 SBC 4

Cluster C6

A

Create an Alpha User cluster for testing Features


Benefits of SME Architecture

Centralized Dial Plan

Can use SIP Trunks between cluster

Multiprotocol

Move 3rd party systems out of cluster

Fax

Turrets

Contact Center

PBX

Better dial plan management and interface then GK/SME

Reduced number of SIP trunks on cluster

108


PSTNSBC SBC

Cluster C5

A

Cluster C4

A

PBX

Turrets

Contact Center

H.323

Cluster C3

A

Cluster C2

A

SME

AMGCP

SIP

SIP

SIP SIP

SIP

SIP

SIP

SBCSBC

SBCSBC

SIP SIP

SME Architecture

PBX

H.323

Cluster C6

A

SBCSBC

IVR

SBCSBC

Cluster C7

A

Cluster C1

A

Cluster C8

A

FAX

SIP

SIP

SIP

SIP

SIP

Covered on BRKUCC2450SME Design

110


Summary

Design challenges

Flat dial plan without access code for PSTN

Possible issues and workarounds

Sip Normalization

Test on lab and pilot until you break something

111


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