Designing Session Management for SIP Trunking and Dial ...d2zmdbbm9feqrf.cloudfront.net/2012/anz/pdf/BRKUCC-2064.pdf · oRun On All Unified CM Nodes In conjunction with the “Route

BRKUCC-2064

Designing Session Management for SIP Trunking and Dial Plan Centralisation

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© 2012 Cisco and/or its affiliates. All rights reserved. Cisco Public BRKUCC-2064 2

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Session Objectives

At the end of this session you should :

Have a good understanding of how to design SME deployments

Understand the benefits of SME designs

Understand how to size an SME cluster

SME DEPLOYMENT GUIDE

The content of this presentation is based on the SME Deployment Guide which can be found here : http://www.cisco.com/en/US/products/ps10661/products_implementation_design_guides_list.html

http://www.cisco.com/en/US/products/ps10661/products_implementation_design_guides_list.html



Agenda What is a Session Management Edition (SME) cluster ?

Trunk Aggregation and Trunk Design recommendations

Call Re-routing in SME deployments

SME Deployments – Secure Trunks – Encrypted Signalling and Media

Dial Plan Aggregation and Number Transformation

Call Admission Control within SME based UC networks

Signalling Delay considerations - Regional SME clusters

QSIG in SME Deployments

Mobility and SME

UC Applications and SME

SME Deployments with Unified Contact Centre Enterprise

Sizing the SME cluster

Summary


What is a Session Management Edition Cluster ?

A CUCM cluster and SME cluster use exactly the same software

A CUCM cluster is typically used to register 10,000s of Phones

An SME cluster is typically used as a platform for Trunk and Dial Plan aggregation

Both CUCM and SME support Voice, Video and Encrypted calls

Support for SME deployments was introduced with UC version 7.1(2)

UC version 8.5 introduces a number of features that enhance SME functionality :

Improved SIP Trunk and H323 Inter Cluster Trunk functionality

Improved through-cluster routing

SIP Normalisation and Transparency scripts

H323 Trunk

MGCP Trunk

SIP Trunk

CUBE CUBE

CUBE


CUBE CUBE

PSTN

H323 MGCP SIP

Unified CM Session Management Edition Cluster

Leaf Unified CM Clusters/ Leaf UC Systems

CUCM Clusters with

H323 QSIG Trunks to SME

PBXs/ CUCM Clusters with SIP Trunks to

SME

CUCM Clusters with

H323 Trunks to SME

PBXs with MGCP Q931 Gateway

Trunks

PBXs with MGCP QSIG

Trunks to SME

CUCM/PBX with SIP QSIG Trunks

to SME

Why deploy Session Management Edition ?

An SME cluster can interconnect 1000’s of UC systems using SIP, H323, or MGCP Trunks SME allows you

to reduce UC system complexity by centralising your dial plan and call routing rules in the SME cluster which

in turn allows you to simplify the dial plan and management of the connected UC systems

Voicemail System

Conferencing System

An SME cluster at the core of your network allows you to flexibly manage your UC system as it grows and

changes…….


Reasons for deploying an SME cluster

CUBE CUBE

PSTN

H323 MGCP SIP


CUCM Clusters with


PBXs/ CUCM Clusters with SIP Trunks to

SME

CUCM Clusters with

H323 Annex M1 Trunks to SME

PBXs with MGCP Q931 Gateway

Trunks

PBXs with MGCPTrunks to SME

CUCM/PBXs with SIP QSIG Trunks to

SME

Voicemail System

Conferencing System

A centralised Dial Plan in SME allows you to :

Globalise and Normalise the called and calling numbers used by all Leaf systems

Manage overlapping number ranges in Leaf systems

Configure “find me” call routing using Route List and Route Groups

Re-route calls via the PSTN when the device cannot be reached via and IP path

SME Mobility features allow you to offer Single Number Reach functionality to devices on 3rd Party UC

systems

SME Normalisation scripts allow you to modify any inbound or outbound SIP message and SDP body content

– simplifying interoperability with 3rd party UC systems


PSTN PSTN PSTN PSTN PSTN PSTN

Phase 1 – PBX Aggregation

PSTN

CUBE CUBE

Phase 2 – Centralised IP PSTN

PSTN

CUBE CUBE

Phase 3 – PBX decommissioning

PSTN

CUBE CUBE

Phase 4 – CUCM cluster

SME cluster - PBX aggregation - SME migration to

CUCM

Trunk Aggregation and Trunk Design recommendations for SME deployments


Trunk Aggregation and Trunk Design recommendations

Summary deployment recommendations for SME Trunks

Inter Cluster Trunk feature comparison

New Trunk features that simplify SME designs

Load Balancing, availability and redundancy

SME Trunks to IOS gateways and 3rd Party UC systems

CUCM Leaf cluster Trunks – UC version design considerations

SME Clustering Over the WAN – design considerations


Summary design and deployment recommendations for SME Trunks

UC version 8.5+ is recommended for SME clusters as it introduces a range new Trunking features that simplify SME deployments

UC version 8.5+ is preferred for CUCM Leaf clusters as this allows them to use the new Trunking features in this release, but earlier versions are also supported

With Leaf CUCM cluster UC version 8.5+ SIP ICTs and Trunks are preferred

With Leaf CUCM cluster UC versions prior to 8.5 – H323 ICTs are preferred

Although MGCP Trunks to gateways offer benefits in terms of ease of dial plan configuration – they do not (cannot) benefit from the SIP and H323 Trunk features introduced with UC version 8.5

© 2012 Cisco and/or its affiliates. All rights reserved. Cisco Public BRKUCC-2064 12 Legend: Yes Limited support

H.323 (Q.SIG) SIP (QSIG)

Support for “+” character

Signalling Authentication and Encryption TLS

Media Encryption

“Run On All Nodes” feature

“Up to 16 destination addresses” feature

Calling Line ID / Name – Presentation / Restriction

Connected Line ID / Name – Presentation / Restriction

OPTIONS Ping

iLBC, AAC, ISAC and G.Clear Support

G.711, G.722, G.723, G.729 Support

SIP Subscribe / Notify, Publish – Presence

QSIG Path Replacement

QSIG Call Completion – No Reply / Busy Subscriber

Topology Aware - RSVP Based Call Admission Control

Message Waiting Indicator (On /Off)

Video / T.38 Fax support

No

Unified CM 8.5 - Inter Cluster Trunks SIP Trunks vs H.323 Trunks – Feature Comparison


New Trunk features that simplify SME designs UC version 8.5 introduces the following Trunk features :

H323 Inter Cluster Trunks

oRun On All Unified CM Nodes

oUp to 16 Destination IP addresses

SIP Trunks and SIP Inter Cluster Trunks


oUp to 16 Destination IP addresses

oSIP Options Ping

oSIP Normalisation Scripts

Route Lists


In conjunction with the “Route Local” feature – these new features greatly

simplify Leaf CUCM cluster and SME cluster Trunk design and deployment


New Trunk features that simplify SME designs

Run On All Unified CM Nodes

Up to 16 Destination IP addresses

These features :

o Reduce the number of configured Trunks

o Simplify call distribution

o Simplify call routing through Leaf and SME clusters by taking advantage of the “Route Local” rule…..

CUBE

CUBE

SIP Trunk A

SIP/H323

ICT Trunk

CUCM SME

CUBE

CUBE

SIP Trunk A

SIP/H323

ICT Trunk

SIP Trunk B

CUCM SME


New Trunk features that simplify SME designs Outbound SIP Trunks, H323 Inter Cluster Trunks and Route Lists can take

advantage of the Route Local Rule by using the “Run On All Unified CM Nodes” feature

The Route Local Rule

If the CUCM node that the inbound call arrives on – also has an instance of the selected outbound trunk for that call – then use this node to onward route the call

The Route Local rule reduces (and can eliminate) call set up traffic between CUCM nodes with a cluster

CUBE

CUBE

SIP Trunk

SIP/H323

ICT Trunk

CUBE

CUBE

SIP Trunk A

SIP/H323

ICT Trunk

Route List

SIP Trunk B


OPTIONS Ping is activated on a per SIP Trunk basis

Each node running the SIP Trunk daemon in the originating cluster uses OPTIONS

Ping to determine the availability of each defined destination IP address

- CUCM will not attempt to establish a call to an unavailable remote peer

- SIP Trunk - “In Service” whilst one remote peer is reachable

- SIP Trunk - “Out Of Service” state when all remote peers are unreachable

- CUCM 8.5 SIP Trunks – Dynamic reachability detection

- Pre CUCM 8.5 Trunks - Per call time out

SIP Trunk

SIP ICT Trunk

UC 8.5 SIP Trunk Feature - OPTIONS Ping

CUBE

CUBE


Unified CM Trunks – Load Balancing, Availability & Redundancy – Summary

UC 8.5

Features

HA Features affecting Calls originating from a CUCM cluster HA Features for Trunk

destinations

Calls over single Trunks Calls over multiple Trunks

Route Lists and Route Groups Up to 16 Destination IP Addresses

H323 ICTs and SIP Trunks

Call Manager Groups

Call Manager Groups

H323 Gatekeeper

DNS SRV – SIP Trunks

Run on All Nodes

H323 ICTs and SIP Trunks

Run on All Nodes

H323 ICTs, SIP Trunks and

Route Groups

SIP OPTIONS Ping

SIP Trunks only

Route Local Route Local


Normalisation Script

Remove “display name” from SIP Header

function fixInboundPai()

local pai = Sip.getHeader("P-Asserted-Identity")

local displayname = getDisplayName(pai)

local uri = getUri(pai)

local number = getUserPart(uri)

if displayname == number

then Sip.modifyHeader("P-Asserted-Identity", uri)

Normalisation allows incoming and outgoing SIP

messages to be modified on their way through a

CUCM SIP Trunk.

The Normalisation feature is designed to improve

interoperability between CUCM SIP Trunks and SIP

based 3rd Party SIP PBXs, Applications & IP PSTN

services.

Normalisation is independent of what the SIP Trunk

connects to on the other side of CUCM. e.g.

• SIP SIP Trunk calls

• Skinny SIP Trunk calls

• H.323 SIP Trunk calls

• MGCP SIP Trunk calls

Normalisation uses a scripting environment to allow

customers to modify SIP messages and SDP

content on a per trunk basis.

IP PSTN MGCP Trunk SIP Trunk

SIP Trunk SIP Trunk

Script

SIP Trunk SIP/SCCP

Script

H323 Trunk SIP Trunk

Script

IP PSTN

UC 8.5 SIP Trunks – Normalisation Scripts

Scripting Guide at : http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/sip_tn/8_5_1/sip_t_n.html


Transparency Script

Sip.allowHeader(“A-Callid”)

Sip.allowHeader(“A-ConversationId”)

function A.inbound_INVITE

Sip.passThroughHeader(“A-Callid”)

Sip.passThroughHeader(“A-ConversationId”)

Sip.passThroughHeaderValue("Supported", "x-nortel-sipvc")

Sip.passThroughUriParameters("From", "uriparm1")

Sip.passThroughHeaderParameters("From", "hparm1",

"hparm2")

end

Transparency allows CUCM to pass headers,

parameters, and SDP content from one SIP call

leg to the other.

The Transparency feature is designed to improve

the operation of and interoperability between 3rd

Party SIP PBXs and Applications connected via

CUCM/SME.

Transparent pass through is only applicable when

the call through CUCM is from SIP Trunk to SIP

Trunk.

• SIP Trunk SIP Trunk calls

Transparency uses the same scripting as

Normalisation to allow customers to pass SIP

messages through CUCM. Transparency and

Normalisation features can be combined.

SIP Trunk SIP Trunk

Script

SIP Trunk SIP Trunk

Script

SIP Trunk SIP Trunk

Script

UC 8.5 SIP Trunks – Transparency Scripts

Scripting Guide at : http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/sip_tn/8_5_1/sip_t_n.html


SIP Trunks vs H323 and MGCP Trunks to gateways

SIP Trunks support the “Run On All Unified CM Nodes” and “Up to 16 destination IP addresses” features

H323 and MGCP Trunks to gateways and 3rd party UC systems support standard Call Manager Groups and a single destination IP address

Using standard Call Manager Groups (rather than Run on All Nodes) increases call set up traffic between nodes within a cluster

Multiple Trunks may be required with the single destination IP address limitation

Note – MGCP Trunks are only active on one node in the Call Manager Group (as the signalling channel is back hauled to CUCM)

H323

ICT Trunk H323 Trunk A

H323 Trunk B

Selected outbound Trunk

Route List

SIP

ICT Trunk MGCP Trunk A

MGCP Trunk B

Selected outbound Trunk

Route List


H.323 SIP MGCP Centralised Provisioning

QSIG Tunneling

Centralised CDR (DS0 Granularity in Unified CM CDR)

MLPP (Preemption)

Hook-flash Transfer with Unified CM

ISDN Overlap Sending No GK

NFAS

SRTP (Unified CM to GW)

CUCM 8.5 “Run On All Nodes” feature 3 Active Nodes in a CMG

1 active Node in a GMG

CUCM 8.5 “Up to 16 destination addresses” feature

Mobility Manager VXML-Based Voice Profile Mgmt

OPTIONS Ping

TCL/VXML Apps (e.g. for CVP Integration)

Voice & Data Integrated Access

Fractional PRI Workaround

TDM Variations: A-DID, E&M, PRI NFAS, CAMA, T1 FGD

ISDN Video Switching on GW

Set numbering Plan Type of Outgoing Calls

G.Clear (Clear Channel Data) Support

H.320 Video

IOS Gateway Trunks SIP Trunks , H.323 Gateways, MGCP Gateways – Feature Comparison

Legend: Yes Limited support No CMG – Call Manager Group


SIP ICT Trunks - Voice, Video and Encryption supported

OPTIONS Ping, SIP Delayed Offer, Run on All Nodes, Multiple Destination Addresses, QSIG over

SIP. (Call Back & Path Replacement)

SIP IP PSTN Trunks – Typically Voice only – Early Offer usually required by SP

OPTIONS Ping, SIP Delayed Offer, If EO required - CUBE provides SIP DO to EO, Run on All

Nodes, Multiple Destination Addresses. (EO should be sent by the SP)

SIP to 3rd Party UC Systems – Typically Voice – Video & Encryption also supported

OPTIONS Ping, SIP Delayed Offer, Run on All Nodes, If required - QSIG over SIP, If end device is

capable it should send Early Offer.

SME Trunk Design Considerations Multi-Cluster Designs – with CUCM 8.5 Leaf Clusters

IP PSTN

CUBE CUBE SIP Early Offer

SIP Delayed Offer

CUCM 8.5 SME

CUCM 8.5

CUCM 8.5

SIP

CUBE SIP DO to EO


Pre 8.5 Clusters - H323 ICT Trunks - Voice, Video and Encryption supported

H323 Slow Start, Call Manager Groups, 3 Destination Addresses per Trunk, QSIG over H323.


OPTIONS Ping, SIP Delayed Offer, If EO required - CUBE provides SIP DO to EO, Run on All


SIP to 3rd Party UC Systems – Typically Voice – Video & Encryption also supported

OPTIONS Ping, SIP Delayed Offer, Run on All Nodes, If required – QSIG over SIP, If end device

is capable it should send Early Offer.

IP PSTN

CUBE CUBE

CUCM 8.5 SME

CUCM 8.0

CUCM 7.X

SIP Early Offer

SIP Delayed Offer

H323 Slow Start

SIP

H323 CUBE SIP DO to EO

SME Trunk Design Considerations Multi-Cluster Designs – with Pre 8.5 CUCM Leaf Clusters


Pre 8.5 Leaf Clusters – H323 ICT Trunks – Voice, Video and Encryption supported

H323 Slow Start, Call Manager Groups, 3 Destination Addresses per Trunk, QSIG over H323.

8.5 Leaf Clusters – SIP ICT Trunks – Voice, Video and Encryption supported

SIP Delayed Offer, OPTIONS Ping, Run on All Nodes, Multiple Destination Addresses, QSIG over

SIP.


OPTIONS Ping, SIP Delayed Offer, If EO required – CUBE provides SIP DO to EO, Run on All


IP PSTN

CUBE CUBE

CUCM 8.5 SME

CUCM 8.0

CUCM 8.5

SIP Early Offer

SIP Delayed Offer

H323 Slow Start

SIP

H323 CUBE SIP DO to EO

SIP Trunk Design Considerations Multi-Cluster Designs – Mixed Leaf Cluster CUCM Versions


Clustering Over the WAN – Calls from Leaf Clusters to SME

In each Leaf Cluster :

• Create and prioritise multiple SIP Trunks in Route Lists to distribute calls to each group of

• SME nodes in each Data Centre, Run Route Lists on all Nodes

• Enable “Run on all Nodes” on each Leaf Cluster SIP Trunk.

• Define destination IP addresses per Trunk for geographic call distribution

• Use SIP Delayed Offer (DO) on all Trunks (reduces MTP usage)

• For IP PSTNs - If Early Offer (EO) is required – Use CUBE SIP DO to EO feature

IP PSTN

CUBE CUBE SIP Delayed Offer

CUCM 8.5 CUCM 8.5

SIP

IP PSTN

CUBE CUBE

San Jose New York West Coast DC

East Coast DC

Primary Secondary

SME 8.5

CUCM 8.5

San Francisco

CUBE SIP DO to EO

SIP Trunk Design Considerations

COW Bandwidth requirements as per

the UC SRND - 80mS RTT


In the SME Cluster :

• Use standard Call Manager Groups for each SIP Trunk – Define destination IP addresses

• for every call processing node in the Leaf cluster

• For routes to each Leaf cluster - Add SIP Trunks to a Route Group and Route List.

• Run Route Lists on all Nodes. Use Local Route Groups to route outbound calls over

• Trunks in the same data centre as the inbound Trunk call

• Use SIP Delayed Offer (DO) on all Trunks (reduces MTP usage)

• For IP PSTNs - If Early Offer (EO) is required – Use CUBE SIP DO to EO feature

IP PSTN

CUBE CUBE SIP Delayed Offer

CUCM 8.5 CUCM 8.5

SIP

IP PSTN

CUBE CUBE

San Jose New York West Coast DC

East Coast DC

West West

SME 8.5

CUCM 8.5

San Francisco

CUBE SIP DO to EO

Clustering Over the WAN – Calls from SME Cluster to Leafs


COW Bandwidth requirements as per

the UC SRND - 80mS RTT

Call Re-Routing in SME deployments


IP PSTN

CUBE

No Circuit Available

x

SME

IP PSTN

CUBE

SME

Call Admission Control

Out Of Bandwidth

x

Call Re-routing in SME Deployments

Single outbound SME Trunk

Call Progress failure messages (e.g. No circuit, Number Unavailable, No bandwidth

etc) are passed back to the originating Leaf cluster – which can then re-route the call


IP PSTN

CUBE CUBE

No Circuit Available

x

SME

IP PSTN

CUBE CUBE

Call Admission Control

Out Of Bandwidth

x

x SME

Call Re-routing in SME Deployments

Multiple outbound SME Trunks in a Route List

Call Progress failure messages (e.g. No circuit, Number Unavailable, No bandwidth

etc) are passed back to the SME cluster – which can then re-route the call

Secure Trunks Encrypted Media and Signalling


SME Deployments – Secure Trunks – Encrypted Signalling and Media

Secure Signalling and encrypted Media can be configured for SIP and H323 Trunks

SIP Trunk security using TLS is preferred over H323

- H323 uses IPSEC and requires IPSEC Tunnels to be set up in the network rather than on CUCM/SME nodes

Using central Certificate Authority with SIP Trunks reduces configuration overhead and simplifies certificate management

Media Encryption (SRTP) can be set up without secure signalling but in this case security keys are sent in the clear


Certificate Authority (CA)

Leaf Cluster A

Session

Management

Edition

Cluster

SIP Inter Cluster Trunk using TLS

for signalling encryption

CA Certificate

CA Signed CUCM

Certificate

Encrypted

Media

Leaf Cluster B

Secure SIP Trunks using TLS and an external CA


Leaf Cluster A

Encrypted

Media

IPSEC Tunnel

Leaf Cluster B

H323 Inter Cluster Trunk using router

based IPSEC for signalling encryption

Session

Management

Edition

Cluster

Secure H323 Trunks using router based IPSec Tunnels




Basic Principles

Number Transformation options

Dial Plan and dialing habit variations

Number Transformation capabilities by Trunk protocol type


Leaf Cluster Leaf UC system SME Cluster

In Leaf clusters/ Leaf UC systems a localised dial plan may be

implemented and users may have local dialing habits

In the SME cluster a unique globalised dial plan should be implemented – Inbound

number globalisation and outbound number localisation can be implemented in

the SME cluster

SME Dial Plan Aggregation – Basic Principles


SME Cluster

With Session Management Edition and CUCM, normalisation of calling and called party information can be

achieved by using a variety of tools :

• Transformation calling search spaces on trunk or device pool level

• Transformations on route patterns

• Transformations on translation patterns

• Transformations on route lists

This presentation focuses on number transformations at the Trunk level......

But first lets look at some variations on dial plan and dialling habit implementations........

Inbound number globalisation and outbound number

localisation can be implemented in the SME cluster

SME - Number Normalisation Tools


Caller (444-4000) dials 8-555-5000

Cluster 1 DNs

444-4XXX Cluster 2 DNs

555-5XXX

Calling Number +44-404-444-4000

Called Number 8-555-5000


Called Number +55-505-555-5000




Called Number 555-5000

Leaf Cluster 1

Internal Directory Numbers (XXX-YYYY)

Abbreviated Dialing (8-XXX-YYYY)

Leaf Cluster 2



SME Cluster

+ E.164 Dial Plan

Internal Directory Numbers (DNs), +E.164 SME Dial

Plan, Abbreviated On Net Dialing


Caller (444-4000) dials 8-555-5000

Cluster 1 DNs

444-4XXX Cluster 2 DNs

555-5XXX

Calling Number 444-4000

Called Number 8-555-5000



Calling Number 8-444-4000


Calling Number 8-444-4000


Leaf Cluster 1



Leaf Cluster 2



SME Cluster

+ E.164 Dial Plan

Internal DNs, +E.164 SME Dial Plan, Abbreviated On

Net Dialing - All number transformations performed on SME cluster


Caller (+44-404-444-1000) dials 55-505-555-5000

Cluster 1 DNs

+44-404-444-4XXX Cluster 2 DNs

+55-505-555-5XXX









Leaf Cluster 1

+ E.164 Directory Numbers

E.164 Dialing

Leaf Cluster 2

+ E.164 Directory Numbers

E.164 Dialing

SME Cluster

+ E.164 Dial Plan

+E.164 DNs, +E.164 SME Dial Plan, E.164 Dialing


SME /CUCM

Cluster

SIP Trunks – Inbound and Outbound

Digit manipulation capabilities

Inbound SIP Trunk Outbound SIP Trunk

Inbound Calling Search Space

AAR Calling Search Space

Prefix DN – Prefix Digits to Incoming Called

Number

Incoming Calling Number – Prefix, Strip

Transform Digits

+ char – carried in SIP

+ char – carried in called and calling number

QSIG IEs

Outbound Called Party Transformation CSS

Outbound Calling Party Transformation CSS

Caller ID DN – Overwrite/Prefix Digits to Outbound

Calling Number

+ char – carried in SIP

+ char – carried in called and calling number QSIG

IEs


SME /CUCM

Cluster

H323 Trunks and Gateways– Inbound and Outbound


Inbound H323 Trunk Outbound H323 Trunk

Inbound Calling Search Space (CSS)



Number


Transform Digits

Incoming Called Number – Prefix, Strip

Transform Digits

+ char – not carried in H323

+ char – not carried in QSIG




Calling Number

+ char – not carried in H323



SME /CUCM

Cluster

MGCP Trunks – Inbound and Outbound


Inbound MGCP Trunk Outbound MGCP Trunk

Inbound Calling Search Space



Number


Transform Digits

+ char – not carried in Q.931





Calling Number

+ char – not carried in Q.931


Call Admission Control (CAC) in SME based deployments


Call Admission Control in SME based UC networks

CAC Options :

Locations based CAC

RSVP based CAC

Combinations of RSVP and Locations based CAC

Local RSVP can be deployed in the SME cluster only

This has advantages in deployments where partially meshed international WAN circuits are used to interconnect SME and Leaf clusters, or where sites are dual homed to the WAN

RSVP can be used to accurately and efficiently use available WAN bandwidth


Call Admission Control in SME based UC networks

Both Locations based Call Admission Control (CAC) and RVSP based CAC can be used in Leaf clusters and the SME cluster…..

Locations based CAC for campus clusters provided by SME

Locations CAC for Leaf clusters and SME using “hub none” locations

Locations CAC for Leaf clusters and SME using “phantom” locations

Local RSVP in the SME cluster

Local RSVP in Leaf and SME clusters

End to End RSVP in Leaf and SME clusters

Mixed End to End RSVP in Leaf and SME clusters and Locations based CAC in other Leaf clusters


Leaf 1

SME ≥ 8.5

Leaf 2

Leaf 3

SIP Trunk

Registered Endpoint

Location Association

LOCATION 1

Associated to Trunks for Leaf 1 and Leaf 2 (both

Leaf Clusters have endpoints located in the same

physical site)

LOCATION 2

Associated to Leaf 3 Trunk (Leaf 3 is a single site

cluster deployment)

SME cluster provides Locations based CAC for

campus clusters


Central Site

Cluster 1

ICT-SME

Hub_None

Location F

- 80 kbps

Location B

- 80 kbps Branch 1 Cluster 1

Branch 2 Cluster 2

SME

Cluster 1 Locations CAC

80k Deduction for

Location B


80k Deduction for

Location F

WAN

ICT-CL1 Hub_None

Central Site

Cluster 2

ICT-CL2 Hub_None

ICT-SME

Hub_None

SIP Trunk

Call Signalling


Media

Multi Branch Leaf clusters use Locations CAC – ICTs in Location

“Hub-None”


Central Site

Cluster 1

ICT-SME

Hub_None

Location F

No WAN

calls

Location B


Branch 2 Cluster 2

SME

Cluster 1

Locations CAC

160k Deduction for

Location B (2 Calls)

Cluster 2

Call Transferred to Cluster

1 Location B

WAN

ICT-CL1 Hub_None

Central Site

Cluster 2

ICT-CL2 Hub_None

ICT-SME

Hub_None

SIP Trunk

Call Signalling


Media

Multi Branch Leaf clusters - Locations CAC – Call

Transfer

Transfer


Central Site

Cluster 1

ICT-SME

Phantom

Location F

- 80 kbps

Location B


Branch 2 Cluster 2

SME


80k Deduction for

Location B


80k Deduction for

Location F

WAN

ICT-CL1 Phantom

Central Site

Cluster 2

ICT-CL2 Phantom

ICT-SME

Phantom

SIP Trunk

Call Signalling


Media

Multi Branch Leaf clusters - Locations based CAC – ICTs in Location

1 2

“Phantom” Phantom Locations allow the

originating location of each call

to be sent over ICTs


Central Site

Cluster 1

ICT-SME

Hub_None

Location F

No WAN

calls

Location B

No WAN

calls Branch 1 Cluster 1

Branch 2 Cluster 2

SME

Cluster 1

Locations CAC

0 kbps Deduction for

Location B

(0 WAN Calls)

Cluster 2

Call Transferred to Cluster

1 Location B

WAN

ICT-CL1 Hub_None

Central Site

Cluster 2

ICT-CL2 Hub_None

ICT-SME

Hub_None

SIP Trunk

Call Signalling


Media

Multi Branch Leaf clusters - Locations CAC – Call Transfer

Transfer

4 3

1 2

Phantom Locations allow the

originating location of each call

to be sent over ICTs

The originating location of the

inbound transferred call is

returned to cluster 1


SME cluster – Deploying Local RSVP CAC

for partially meshed WANs

Local RSVP in the SME cluster has advantages in deployments where partially meshed international

WAN circuits are used, or where sites are dual homed to the WAN. RSVP does not need to be

implemented in the leaf clusters

For CUCM Leaf Clusters

Using Locations based Call Admission Control

What bandwidth value should be used for calls over SIP inter cluster Trunks into the WAN ?

For Voice Gateways

Using Call Counting based Call Admission Control

What value should be used for calls over SIP inter cluster Trunks into the WAN ?

IP PSTN

CUBE CUBE

WAN Link

UC Bandwidth

10

10 10

10 1 1

10

100


Local RSVP Locations Based CAC Local RSVP

Leaf Cluster 2

7.X or below

SME ≥ 8.5

Leaf Cluster 1

7.X or below Location

Leaf 1

Location

Leaf 2 Inter Location

RSVP Policy =

Mandatory

SME cluster – Local RSVP CAC

RSVP Agents registered to Leaf and

SME clusters may be co-located on

the same platform

SME manages RSVP Agents

between Clusters

SCCP

H323 Inter cluster Trunk

Local RSVP in the SME cluster has advantages in deployments where partially meshed international WAN

circuits are used, or where sites are dual homed to the WAN. RSVP does not need to be implemented in

the leaf clusters


Local RSVP Locations Based CAC Local RSVP

Leaf Cluster 2

7.X or below

SME ≥ 8.5

Leaf Cluster 1


Leaf 1

Location

Leaf 2



the same platform

SME cluster – Local RSVP CAC – Inter cluster Call Setup

SME RSVP

Reservation

Call Setup 1

2

3 Call Setup

Call Signalling

SCCP


RSVP

Leaf

RSVP

Resv.

With Local RSVP in the SME cluster – Call set up proceeds on the outbound SME Trunk once the RSVP

Reservation has succeeded between the two SME controlled RSVP agents


End-to-End RSVP with SIP Preconditions

Leaf Cluster 2

8.0 or above

SME ≥ 8.5

Leaf Cluster 1

8.0 or above Location

Leaf 1

Location

Leaf 2

SME cluster – End to End RSVP CAC with SIP

Preconditions

SME RSVP

Reservation

SCCP

SIP Inter cluster Trunk

RSVP

SIP Preconditions passed over SIP Inter cluster

Trunks from Cluster 1 via SME to Cluster 2


Local RSVP and End-to-End RSVP with SIP Preconditions

Locations Based CAC

Leaf Cluster 2

8.0 or above

SME ≥ 8.5

Leaf Cluster 1


Leaf 1

Location

Leaf 2 Inter Location

RSVP Policy =

Mandatory

SME cluster – End-to-End RSVP with SIP

Preconditions extended into one Leaf cluster

SCCP


SIP Inter cluster Trunk

SME RSVP

Reservation


Local RSVP Local RSVP

Leaf Cluster 2

SME ≥ 8.5

Location

Leaf 2

SME cluster – Local RSVP CAC – Co-Locating RSVP

agents on the same router platform

SME and Leaf cluster RSVP

Agents co-located

on the same router platform

sccp local GigabitEthernet0/0

sccp ccm 10.10.10.10 identifier 1 version 8.0

sccp ccm 10.10.20.20 identifier 2 version 7.0

sccp ip precedence 3

sccp

…

sccp ccm group 10

associate ccm 1 priority 1

associate profile 10 register RSVP_Agent_SME

…

sccp ccm group 20

associate ccm 2 priority 1

associate profile 20 register

RSVP_Agent_Leaf2

…

dspfarm profile 10 mtp

codec g729r8

codec pass-through

rsvp

maximum sessions software 300

associate application SCCP

…

dspfarm profile 20 mtp

codec g729r8

codec pass-through

rsvp

maximum sessions software 300

associate application SCCP



the same platform

Signalling Delay Considerations Regional SME Clusters


SME

Recommendations for media delay are well defined

(ITU Recommendation G.114. < 150mS = acceptable, 150 – 400mS = acceptable with some impact on

quality, > 400mS generally unacceptable)

Recommendations for signalling delay are not well defined

Primarily because the incurred delays are protocol dependent and the impact of long delay generally

affects call set up rather than overall voice quality

Signalling Delay and SME based UC networks


Two Way Media

INVITE

200 OK w/ SDP (Offer)

ACK w/ SDP (Answer)

180 Ringing

100 Trying

INVITE


ACK w/ SDP (Answer)

180 Ringing

100 Trying

Messages exchanged before called user

hears the caller after picking up their

handset

Messages exchanged before the caller

hears ringback tone

Delay before the caller hears the called

user after ringback stops

One way signalling Delay

SME

Impact of Signalling Delay on Call Set Up in SME networks

The diagram above shows an example of call set up delays and their impact on the users’

experience. (Note – Phone to Call Agent signalling delay has been assumed to be minimal)

Delays during call set up will vary based on the protocol(s) used, the trunk configuration and call

agent operation – making it difficult to calculate the time taken to establish each stage of the call

set up.

In most cases, signalling delays do not noticeably affect user experience. If signalling delays are a

concern, consider deploying regional SME clusters.


IP PSTN

CUBE CUBE

SME

America

IP PSTN

CUBE CUBE

SME

Europe

IP PSTN

CUBE

SME

Asia Pac

Regional SME clusters

Provide a Regionalised SME cluster if there are multiple leaf UC systems in a geographic region with

significant amounts of intra region traffic

If regionalised SME clusters are not deployed, situate your SME cluster(s) closest to those leaf UC

systems that generate the most inter cluster traffic

QSIG and SME deployments


QSIG and SME UC deployments With UC 8.5 - QSIG is supported over SIP, H323 and MGCP Trunks

QSIG specific Features :

oCall Back on Busy

oCall Back on No Answer

oPath Replacement

oCalling Name

oConnected Name

QSIG features such as Call Back and Path Replacement can use calling and called numbers to determine whether the feature should be invoked. Numbers carried in QSIG APDUs are not modified by CUCM number transforms. In SME designs where number normalisation is deployed – Use the following CUCM/ SME service parameters :

oCall Back : Connection Retention (default setting)

oPath Replacement : PINX ID – Use a globally unique number for the PINX ID of each cluster (The PINX ID is equivalent to any routable number in the UC system and uniquely identifies a cluster)


Phone 1000 calls Phone 4000

- Signalling Call leg from 1XXX cluster via SME to 4XXX cluster

- Call Admission Control deducts the bandwidth for a call from 1XXX to 4XXX

- RTP media path direct between the two Phones over the WAN

Benefits of QSIG Path Replacement in SME based

UC deployments IP PSTN

CUBE CUBE

10

10

1000

1001

4000 9

9

QSIG over SIP

QSIG over H323

Media


Phone 4000 Transfers the call to Phone 1001

- New Signalling Call leg from 4XXX cluster via SME to 1XXX cluster

- Call Admission Control deducts the bandwidth for a call from 4XXX to 1XXX

- Resulting Media Path direct between Phone 1000 and Phone 1001


UC deployments IP PSTN

CUBE CUBE

10

10

1000

1001

4000 8

8

QSIG over SIP

QSIG over H323

Media


QSIG Path Replacement for Tromboned calls

Clears down redundant signalling

Releases unused Call Admission Control bandwidth

IP PSTN

CUBE CUBE

10

10

1000

1001

4000 10

10

SCCP/SIP

Media


UC deployments

For more info on QSIG and SME deployments see

The SME Deployment Guide :


and the CUCM Admin Guide :

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08procl.html#wp1139777


http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08procl.html

Cisco Mobility and SME deployments


Mobility Features in SME based UC networks

For SME based UC designs - Unified Mobility can be deployed as follows :

Standard Leaf CUCM cluster mobility deployment – where all mobility features and PSTN access are provided on the Leaf cluster

Leaf cluster mobility with SME based PSTN access

3rd Party PBX Mobility/ Single Number Reach support with PSTN access via SME

Mobility features provided by SME for a connected 3rd party PBX

SIP Early Offer

SIP Delayed Offer

H323 Slow Start

IP PSTN

CUBE

SME

CUCM 8.0

PBX with SNR capability

Unified Mobility

User

Unified Mobility

User

PBX using SME based

Mobility features

CUCM 8.5

CUBE

TDM PSTN

SIP

H323


Leaf cluster mobility with SME based PSTN access

Supported Features

Mobile Connect (aka SNR)

Mobile Voice Access (MVA)

Desk Phone pickup via hangup at mobile device and resume at desk phone.

Remote Destination pickup using "Send Call to Mobile"

Dual-mode Phones and Clients / Direct Connect (Cisco Jabber) Mobile Clients / BlackBerry MVS

Unsupported Features - Features that use DTMF tones for feature invocation :

Enterprise Feature Access (EFA)

Desk Phone and Remote Destination pickup

Mid-call Supplementary Services

IP PSTN

CUBE

SME CUCM 8.0

Unified Mobility

User

Unified Mobility

User

CUCM 8.5

CUBE

TDM PSTN

Note – SME PSTN access

can be TDM or IP based


Mobility features provided by SME for a 3rd party

PBX IP

PSTN

CUBE

SME PBX with SNR

capability

Unified Mobility

User

Unified Mobility

User

PBX using SME based

Mobility features

CUBE

TDM PSTN

Where a 3rd Party PBX does not natively support any mobility features, the SME cluster can act as a

mobility feature proxy by creating a Remote Destination Profile and two or more Remote Destination

numbers/ IDs per user. One Remote Destination will be the directory number of the user’s 3rd Party PBX

Phone and an additional Remote Destination number can be the user’s mobile phone number.

Supported mobility features :

Mobile Connect (aka SNR)

Mobile Voice Access (MVA)

Unsupported Features - Features that use DTMF tones for feature invocation :

Enterprise Feature Access (EFA) / Desk Phone and Remote Destination pickup / Mid-call Supp. Services

Note – SME PSTN access

can be TDM or IP based

UC Applications in SME deployments


UC Applications in SME deployments

H323 MGCP SIP

Session Management

Edition Cluster

Voicemail System

Conferencing System

CER Cisco Unified

Contact Centre

Cisco Emergency Responder

Cisco Unified Presence Server

Operator Console

SME supports the following centralised applications :

Unity, Unity Connection

Meeting Place, Meeting Place Express

SIP and H323 based Video Conferencing systems

3rd Party Voice Mail systems

Fax servers

Applications that track user or phone state must connect to the leaf cluster :

Unified Contact Centre, Unified Contact Centre Express


Attendant Console



Centralised Unity VoiceMail in SME deployments

H323 MGCP SIP

Session Management

Edition Cluster

Voicemail System

Conferencing System

CER Cisco Unified

Contact Centre



Operator Console

On Trunks between leaf clusters and SME and on Trunk connections to your Voicemail application ensure that the original

called party/ redirecting number is sent with calls routed to voicemail.

For Non QSIG Trunks – Original Called Party /Redirecting number transport can be enabled by :

Enabling inbound and outbound Redirecting Diversion Header Delivery on SIP Trunks

Enabling inbound and outbound Redirecting Number IE Delivery on MGCP Gateways, H.323 Gateways and H.323 Trunks

For QSIG enabled SIP, MGCP and H323 Trunks – the Original Called Party number is sent in QSIG Diverting Leg

Information APDUs. Note - The + character is not sent with the Diversion information sent in QSIG APDUs.

For more information see the SME Deployment Guide : http://www.cisco.com/en/US/products/ps10661/products_implementation_design_guides_list.html



CUBE CUBE

IP PSTN

EMCC Trunks Inter Cluster Trunks


SME Deployments with Extension Mobility Cross

Cluster

EMCC can be deployed as an overlay between leaf clusters in an SME based UC network.

Leaf cluster EMCC Trunks do not point to the SME cluster.

In each Leaf cluster, an EMCC Trunk(s) and its corresponding Remote Cluster destination information is

defined such that each EMCC Trunk can reach all other EMCC enabled leaf clusters directly.

SME deployments with Cisco Unified Contact Centre


SME Deployments with UCCE

Unified CM SME can be deployed in Unified Contact Centre Enterprise (CCE) designs, where one or more leaf clusters are Contact Centre-Enabled.

There are specific requirements for the architecture used in these scenarios :

• Unified CCE is only certified using SIP as the protocol interconnecting the solution components.

• The preferred DTMF transport in Unified CCE is RFC2833 (“inband DTMF”), since it is the only method supported for self-service applications on Unified CVP

• Unified CCE and Unified CVP often use SIP headers to pass information between systems. By default, SME does not transparently relay some of these SIP headers and payloads from one trunk to another. However, if required, a transparency script can be applied to the SME trunk.

• See the SME Deployment Guide for more details

• http://www.cisco.com/en/US/products/ps10661/products_implementation_design_guides_list.html




JTAPI

Contact Centre CUCM Cluster

VXML Gateway

CUBE

Back Office CUCM’s/PBX’s

Back-office calls

CUSP

Contact Centre calls

IP PSTN

Queued Calls

Calls Transferred to Agents

Inbound calls to Contact Centre flow through CUSP and are routed to a Queue (e.g. On VXML gateway) via CUSP

Calls transferred from the Queue to an Agent are routed through SME and optionally CUSP

Calls to Back Office Users are routed via CUBE and SME No Calls are transferred from Back Office to Contact Centre

SUPPORTED DESIGN



JTAPI

Contact Centre CUCM Cluster

VXML Gateway

CUBE

Back Office CUCM’s/PBX’s

IP PSTN

Queued Calls NOT SUPPORTED Inbound calls to Contact Centre flow through SME and are routed to a Queue (e.g. On VXML gateway)

NOT SUPPORTED

IPv6 based SME deployments


IPv6 based SME Deployments

Cisco UC products supporting IPv4 and IPv6 :

CUCM/SME SIP Trunks

Newer SCCP based Phones

SIP based IOS gateways

Cisco Unified Communications Manager Express and CUBE

SCCP based VG224 Analog gateways and Router FXS ports,

IOS MTPs

SIP Trunks to Unity Connection

Dual Stack (IPv4 and IPv6) configuration is recommended

Dual Stack ANAT is recommended for SIP Trunks http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/ipv6/ipv6srnd.html

CUBE CUBE

PSTN

SIP Unified CM Session Management Edition Cluster

Voicemail System

Conferencing System

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/ipv6/ipv6srnd.html

Centralised and Distributed PSTN


Centralised PSTN

Centralised IP PSTN deployments are becoming increasingly popular as savings can be made through bulk call minute deals with Service Providers and by eliminating TDM circuits in branches

Bear in mind that with no WAN connectivity no PSTN calls can be made or received

A number of analog PSTN ports are often provided in the branch for back up and emergency outbound calls

For emergency calls to the centralised PSTN, MLPP can be used to pre-empt existing WAN calls if insufficient CAC bandwidth is available

CUBE CUBE

PSTN

H323

MGCP

SIP


Voicemail System

Conferencing System


Distributed PSTN

Typically TDM based branch PSTN circuits

Distributed IP PSTN deployments growing

Enables Tail End Hop Off

Can be combined with centralised outbound PSTN on SME or centralised outbound PSTN on Leaf clusters which allows (for example, in country) TEHO to be deployed

H323 MGCP SIP


Voicemail System

Conferencing System

PSTN CUBE

PSTN CUBE

PSTN CUBE

PSTN PSTN PSTN

SME Cluster Sizing


SME Cluster Sizing – Summary Guidance

CUBE CUBE

PSTN

H323

MGCP

SIP


CUCM Clusters with


PBXs/ CUCM Clusters with

SIP Trunks to SME

CUCM Clusters with

H323 Annex M1 Trunks to SME

PBXs with MGCP Q931 Gateway Trunks

PBXs with MGCPTrunks to SME

CUCM/PBXs with SIP QSIG Trunks to SME

Voicemail System

Conferencing System

User Traffic information required to size your SME cluster

Average User Busy Hour Call Attempts (BHCA)

Average Call Holding Time (CHT)

% Off Net Traffic (Off Net to PSTN via SME)

% On Net Traffic

% Intra cluster Traffic

Work with your Cisco account team or partner to size your SME cluster using the SME sizing tool -

http://tools.cisco.com/cucst/faces/login.jsp

http://tools.cisco.com/cucst/faces/login.jsp

Summary


Summary

SME deployments allow you to simplify the UC network edge by aggregating dial plan, services, applications and PSTN access on an SME cluster

SME based UC aggregation simplifies the management of your UC network as it grows and changes

For design guidance see the SME Deployment Guide :




Q & A


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Additional Slides


SIP Trunk Signalling and Basic Operation SIP Messaging - Delayed and Early Offer

SIP Early Offer

Information about the calling device’s media characteristics are sent with its initial SIP INVITE message – The media characteristics are contained in the Session Description Protocol (SDP) body sent with the SIP INVITE – The “Offer” in the SDP body will contain the IP Address, UDP Port number, list of codecs etc. supported by the calling device

The called device selects which of the offered codecs it wishes to use for the call and returns its “Answer” in the SDP body of a SIP response – The Answer also contains the IP address and UDP port number etc of the called device

Once the Answer has been received and acknowledged two way media can be established

Early Offer is widely used …….

Two Way Media

INVITE w/ SDP (Offer)

200 OK w/ SDP (Answer)

ACK

180 Ringing

100 Trying



ACK

180 Ringing

100 Trying



SIP Delayed Offer No information about the calling device’s media characteristics are sent with its initial SIP INVITE message.

Instead the media characteristics are sent by the called device in the Session Description Protocol (SDP) body of the next reliable message (200 OK) – The called device’s “Offer” will contain its IP Address, UDP Port number, list of codecs etc.

The calling device selects which of the offered codecs it wishes to use for the call and returns its “Answer” in the SDP body of a reliable SIP response (ACK) – The Answer also contains the IP address and UDP port number etc of the calling device

Delayed Offer is a mandatory part of the SIP standard – Many SPs prefer Early Offer

Ordinarily, the Offer or Answer cannot be sent with 100 Trying or 180 Ringing as 1XX messages are unreliable (unacknowledged)… This can be resolved using PRACK ….. discussed later…..

Two Way Media

INVITE


ACK w/ SDP (Answer)

180 Ringing

100 Trying

INVITE


ACK w/ SDP (Answer)

180 Ringing

100 Trying



Inbound SIP Delayed Offer to Outbound SIP Early offer

So what happens when Unified CM receives an inbound call on a Delayed Offer Trunk and needs to onward route the call over a Early Offer Trunk ?

It does not have the calling device’s media characteristics and it needs to send an Offer in SDP with the outbound INVITE…

Solution – Insert a Media Termination Point (MTP) and use its media characteristics to create the Offer in SDP with the outbound INVITE

Two Way Media

INVITE


ACK w/ SDP (Answer)

180 Ringing

100 Trying

INVITE w/ SDP (MTP)


ACK

180 Ringing

100 Trying

SIP Delayed Offer SIP Early Offer

MTP


SIP Trunk Signalling and Basic Operation SIP Messaging – Enabling SIP Early Offer – Method 1

SIP Trunk “MTP Required” Checkbox

SIP Early Offer Trunks use the Trunk’s Media Termination Point (MTP) resources, inserting an MTP into

the media path for every outbound call – sending the MTP’s IP Address, UDP port number and codec in

the SDP body of the initial SIP INVITE instead of those of the endpoint.

This has a number of disadvantages : MTPs support a single Audio codec only e.g. G711 or G729. The

passthru codec is not supported excluding the use of SRTP and video calls. Since the Trunk’s MTPs are

used rather than the calling device’s MTPs - The media path is forced to follow the signalling path.

SIP Trunk with Early Offer SIP Line

SCCP Line

SIP Trunk

H323 Trunk

MGCP Trunk

SIP Trunk with Early Offer




MTP

MTP

MTP

MTP

MTP

Using the “MTP Required” option :


New SIP Profile checkbox “Early Offer support for

voice and video calls (insert MTP if needed)”

For Calls from trunks and devices that can provide

their IP Address, UDP port number and supported

codecs - This information is sent in the SDP body of

the initial SIP Invite on the outbound Early Offer

Trunk. No MTP is used for the Early Offer

For Calls from trunks and devices that cannot

provide Early Offer information – use the calling

device’s MTP resources (first) or the outbound

trunk’s MTPs (second) to create a SIP Offer for an

unencrypted voice call. (SRTP and video can

subsequently be initiated by the called device)


SCCP Line SIP Trunk with Early Offer

SIP Trunk SIP Trunk with Early Offer

H323 Trunk SIP Trunk with Early Offer

MGCP Trunk SIP Trunk with Early Offer

SCCP Line SIP Trunk with Early Offer MTP

SIP Trunk SIP Trunk with Early Offer MTP


MTP

Cisco SIP Phones

Newer SCCP Phones

Older SCCP Phones

SIP Early Offer

SIP Delayed Offer

H323 Slow Start

H323 Fast Start

MGCP Gateway


New

Features


Benefits of “Early Offer support for voice and video calls (insert MTP if needed)”

• Reduced MTP usage

• Single voice codec MTP limitation removed (by using the pass through codec)

• Voice codecs sent in SIP Offer based on calling device capabilities & region settings

• Video Calls supported

• Encryption supported

• Use of the Calling device’s MTP rather than Trunk’s MTP

- media does not have to follow the signalling path


SCCP Line SIP Trunk with Early Offer

SIP Trunk SIP Trunk with Early Offer


MGCP Trunk SIP Trunk with Early Offer

SCCP Line SIP Trunk with Early Offer MTP

SIP Trunk SIP Trunk with Early Offer MTP


MTP

Cisco SIP Phones

Newer SCCP Phones

Older SCCP Phones

SIP Early Offer

SIP Delayed Offer

H323 Slow Start

H323 Fast Start

MGCP Gateway


New

Features


SIP Trunk Signalling and Operation – PRACK (1)

SIP Early Media – Using Provisional Acknowledgement (PRACK)

SIP defines two types of responses: Final and Provisional.

Final responses convey the result of the processed request, and are sent reliably (i.e. they are acknowledged).

Provisional responses provide information on the progress of the request, but are not sent reliably – so the sender of a provisional response does know that it has been received.

To send an Offer or Answer with a provisional 1XX response – these responses must be sent reliably…..

PRACK – Provisional Acknowledgement is used to provide 1XX responses with reliability.

Diagram : Early Offer with Early Media

Two Way Media

INVITE w/ SDP Supported:100rel

PRACK

200 OK (PRACK)

183 Progress w/ SDP Require:100rel

100 Trying

INVITE w/ SDP Supported:100rel

PRACK

200 OK (PRACK)


100 Trying


SIP Trunk Signalling and Operation – PRACK (2) SIP Early Media – Using Provisional Acknowledgement (PRACK)

Like final responses, by using PRACK - 1XX messages will be periodically re-sent until their receipt is acknowledged by the receiver by sending a PRACK, which is also acknowledged by the 1XX sender.

Using PRACK can reduce the number of SIP messages that need to be sent before two way media can be established

PRACK is useful in situations where long Round Trip Times between SIP devices can cause a delay to media cut through or media clipping

PRACK can be enabled on the SIP Trunk Profile by setting “SIPRel1XX Options” to enabled

Diagram : Delayed Offer with Early Media

Two Way Media

INVITE Supported:100rel

PRACK w/ SDP

200 OK (PRACK)


100 Trying

INVITE Supported:100rel

PRACK w/ SDP

200 OK (PRACK)


100 Trying


SIP Trunk SIP Trunk

Cluster 1 Cluster 2 Cluster 1 Cluster 2


Using standard Call Manager Groups and multiple

destinations IP addresses

Unified CM SIP Trunks will only accept inbound calls from a device with an IP address

that has been defined as a destination IP address on the Trunk

Cluster 1 – SIP Trunk configuration

Servers A, B and C in SIP Trunk’s Call

Manager Group

Servers D, E and F are configured as

Trunk destinations

A

B

C

D

E

F

A

B

C

D

E

F


Servers D, E and F in SIP Trunk’s Call

Manager Group

Servers A, B and C are configured as

Trunk destinations


SIP Trunk SIP Trunk


Using Run on all Active Unified CM Nodes and multiple

destination IP addresses


The SIP Trunk has an active SIP daemon

on Servers A, B, C and D

Servers E, F, G, H and I are configured as

Trunk destinations



on Servers E, F, G, H and I

Servers A, B, C and D are configured as

Trunk destinations

A

B

C

E

F

G

D

A

B

C

D

H

I

E

F

G

H

I





SIP Trunk SIP Trunk


Using standard Call Manager Groups, Run on all Active Unified

CM Nodes and multiple destination IP addresses A

B

C

D

E

F

G

H

A

B

C

D

E

F

G

H



on Servers A, B, C, D and E

Servers F, G and H are defined as Trunk

destinations


Servers F, G and H in SIP Trunk’s Call

Manager Group

Servers A, B, C, D and E are defined as

Trunk destinations





Two Way Media

INVITE


ACK w/ SDP (Answer)

180 Ringing

100 Trying

INVITE


ACK w/ SDP (Answer)

180 Ringing

100 Trying

Messages exchanged before called

user hears the caller after picking up

their handset

Messages exchanged before the

caller hears ringback tone

Delay before the caller hears the

called user after ringback stops


SME

Signalling Delay – SIP Delayed Offer Trunks


Two Way Media



ACK

180 Ringing

100 Trying



ACK

180 Ringing

100 Trying

Messages exchanged before called

user hears the caller after picking up

their handset

Messages exchanged before the

caller hears ringback tone

Delay before the caller hears the

called user after ringback stops


Signalling Delay – SIP Early Offer Trunks


Documents

Designing Session Management for SIP Trunking and Dial ...d2zmdbbm9feqrf.cloudfront.net/2012/anz/pdf/BRKUCC-2064.pdf · oRun On All Unified CM Nodes In conjunction with the “Route