Brkucc 2735

SIP Trunk Design and Deployment
Playbook for the Enterprise

BRKUCC-2735
vbharga@cisco.com
The Unified CM SIP Trunk
Definition
 SIP Trunks can be defined as a way to interconnect different
SIP-based networks. One common application for SIP Trunks
for enterprises is to obtain IP-PSTN services from service
providers. PSTN services based on the SIP Trunk give
enterprises the ability to aggregate their telephony services
over a combined IP infrastructure, reducing the cost and
complexity of the network and providing a single point of
interconnect to their users
 SIP Trunks are also used within the enterprise to connect SIP
enabled systems together to harness the full power of SIP
based communications and have the advantage of a single full
featured communications protocol
 This session examines the role SIP Trunks play in Cisco’s
Unified Communications applications by first analyzing the UC
Manager SIP implementation and then its interconnections
with other SIP servers
BRKUCC-2735_c1 © 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 2
SIP Trunk in Unified CM 4.x
Cisco Unified Communications
Manager 4.x SIP Network
Gateways
SIP Trunk
Conf/Xcode
DSP Resources
Video
Rich-Media
Endpoints
Conferencing Voicemail
Soft
Unified Phones Gateways
Messaging
CTI Apps Soft
Cisco and 3rd-party Phones Cisco and 3rd-party
Phones SCCP
Phones
MGCP
H.323
Cisco Unified Communications Manager 4.X SIP support limited to
trunk-side interfaces only. Basic audio calls only – no SIMPLE/presence CTI
support, no video, etc. Not recommended for mass-deployment SIP
SIP Line & Trunk in Unified CM 7.x
Cisco Unified Carriers /
Communications Cisco Unified other vendors
Manager 7.x CME PBXs
Gateways
Conf/ Xcode
DSP Resources Manager 7.x
Cisco Unified
Rich-Media Presence
Conferencing
Microsoft
Unified LCS/OCS
Messaging
IBM Lotus
Sametime
Soft Video Cisco Unified
CTI Apps Cisco and
Phones Endpoints Personal SCCP
3rd-party Phones
Communicator MGCP
H.323
Cisco Unified Communications integrates rich, native SIP support on both CTI
line-side and trunk-side interfaces while maintaining seamless inter-working SIP/SIMPLE
with existing H.323, MGCP, SCCP, TAPI/JTAPI and Q.SIG protocols CSTA over SIP
SIP Trunk in Unified CM 8.0 Carriers / other
PBXs/
Enterprises
Cisco Unified Cisco Unified
Communications CME CUCM-SME
Manager 8.0(1)
Gateways

Manager 8.0(1)
SAF-enabled
Rich-Media IP Network
Conferencing
Cisco Unified
Presence
Unified
Messaging
Microsoft
LCS/OCS
CTI Apps DSP Resources
Cisco Unified SCCP
IBM
Personal Lotus MGCP
Communicator Sametime H.323
Cisco Unified Communications extends SIP based integration to new CTI
deployment models, administration, and opens new avenues for inter- SIP/SIMPLE
company collaboration CSTA over SIP
SIP Line vs. SIP Trunk
 Unified CM SIP Line Side

Interface is different than its
SIP Trunk Side Interface
Line side is geared towards the
end user, contains Cisco
extensions, and is full featured
Typical use: Phones
Trunk side is geared towards
the network interface and
supports only a subset of
services. Emphasizes
interoperability
Typical use: Proxies, Other
Unified CM, Session Border
Controllers, Gateways, etc.
SIP Line Interface provides SIP

SIP Trunk Interface provides SIP
based connectivity and features
based connectivity to external
to user facing devices and is
SIP servers
internal to the CUCM Cluster
Agenda
 Technology Basics of the Unified CM SIP Trunk
Operating Parameters
SIP Signaling
Redundancy, Availability and Load Balancing
Codec Support
Media Resource Requirements & Usage
Security
 What’s New in 8.0
Service Types (IME, EMCC, SAF/CCD)
End to End RSVP
Session Management Edition Deployment Model
 SIP Trunk Integration with UC Components
Intercluster
Gateways
SIP Servers: CUBE, CUSP, UCME
Voice Messaging, Conferencing and Collaboration, Video
 Conclusion and Key Takeaways
SIP Trunk Dynamics
Administration->Device->Trunk
 Unified CM SIP Trunk has lots

of moving parts that must be
tuned for optimum
performance
Device Pool determines CM
Subscriber Group
Media Resource Group List
selects the resources usable by
trunk
Select MTP Required to force an
MTP in media path
Destination Address: IP address
or DNS Name or SRV
Use SIP Trunk Security Profile
to define listening port, enable
security settings and other
parameters
DTMF Signaling Method –
choose OOB, RFC-2833, or both
SIP Trunk Dynamics
Administration->System->Security Profile->SIP Trunk Security Profile
 Use the SIP Trunk

Security Profile to set
Incoming and Outgoing
transport types for
transport protocol – UDP,
TCP, or TLS
Digest Authentication to
enable challenge –response
for every request
Incoming port for the port
CUCM will listen for
connections
SIP Messages that the
Administration->Device->Device Settings->SIP Profile trunk will accept
 Use the SIP Profile

Information for
DTMF payload type
and other SIP protocol
specific settings
Unified CM SIP Trunk Technology Basics
SIP Signaling
Redundancy, Availability and Load Balancing
Codec Support
Media Resource Requirements
Security
Presentation_ID © 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 10
Overview of the Unified CM SIP Trunk
SIP Messaging - Delayed and Early Offer
 Why does the Unified CM

do Delayed Offer?
Protocol conversion from SCCP
and H.323 to SIP
INVITE w/ SDP
(Offer) INVITE w/o SDP Determine capabilities and
(No Offer) insert media resources if
needed
180 Ringing 180 Ringing
200 OK w/ SDP  Can it do regular/early offer?
(Offer)
It needs a known IP/Port
200 OK w/ SDP parameters to form an offer.
(Answer) For that it needs a Media
ACK w/ SDP
ACK (Answer) Termination Point (MTP)
 What about media clipping?

Two Way Media
Recommend usage of PRACK
to complete Offer/Answer
exchange before 200 OK from
called party
SIP Messaging - Hold/Resume Signaling
User
Presses
Hold Two Way Media  Unified CM sends
INVITE w/ SDP
(a=sendonly) INVITE w/ SDP
 a=inactive to stop media
(a=inactive)
 delayed offer to start MoH
200 OK w/ SDP
200 OK w/ SDP (a=inactive) or resume media
(a=recvonly) ACK
 Remote entity must respond
ACK
with its full codec list and
INVITE w/ SDP a=sendrecv when it receives
(a=sendrecv) INVITE w/o SDP delayed offer for MoH or to
200 OK w/ SDP
200 OK w/ SDP resume held call
(a=sendrecv)
User (a=sendrecv) ACK w/SDP
Presses
ACK (a=sendrecv)
Resume
Two Way Media
SIP Messaging - Transfers
 Unified CM -
 Places active call on
User
Presses
hold and plays MoH
Transfer
 Connects to transfer
Two Way Media
target
Initiates HOLD
 Updates connected
name/number for
Initiate HOLD
each call
HOLD Complete HOLD Complete
 Updates SDP for
Initiate Second Call each leg
Initiate Second Call
Call Complete  Disconnects
User
Dials Call Complete
2nd
Transferor
Number
REFER  Media flows directly
INVITE/200/ACK between the connected
(with C SDP)
User INVITE/200/ACK parties
Presses 200 OK
Transfer
(with B SDP)  Does NOT use REFER
Again BYE and maintains call
A BYE B C signaling control
A: Transferor
Two Way Media B: Transferee
C: Transfer Target
Audio Codec Determination
Configuration
(64K Audio Inter-region bps)
Originating Side Terminating Side  Codec matching determined
Offer
Offer
(G.711, G.729, G.722) by an intersection of codecs
(G.711, G.723, G.729)
in originating and
destination offers and inter-
region configuration
 Better quality codecs in the
Selection Policies intersection preferred over
lesser (G.711 over G.729)
Outgoing
Answer (G.711)
Video Codec Determination
 Video Bit-rate between and within regions

determines video resolution and frame rate
 Usage of bandwidth parameter in SDP signaling
(TIAS as in TIAS: 38400) forces endpoint selection
of desired resolution and frame rate
Resolution Frame Rate Region Bit Rate Required
QCIF 10 fps 60,000 - 79,999 bps
QCIF 15 fps 80,000 - 99,999 bps
QCIF 30 fps 100,000 - 249,999 bps
CIF 15 fps 250,000 - 299,999 bps
CIF 30 fps 300,000 - 499,999 bps
VGA 15 fps 500,000 - 799,999 bps
VGA 24 fps 800,000 - 999,999 bps
Unified CM default is 384,000 bps (CIF, 30 fps)

Redundancy and Load Balancing — Multiple Sources
 SIP Trunk is defined on one

or more UCM subscribers
 All of these subscribers are
able to originate and receive
calls on the trunk
 Outgoing calls from cluster
are distributed over these
subscribers
 Incoming calls should be
sent to one of these
subscribers only!
 Single trunk only –
redundancy on sending
but not receiving side
Redundancy and Load Balancing — Multiple Sources
Administration->Device->Trunk
Device Pool setting in Trunk Config
Administration->System->Device Pool
Device Pool contains CallManager
Group
Administration->System->Cisco Unified CM Group

CallManager Group contains one or
more subscribers
Redundancy and Load Balancing — Multiple Trunks
 Redundant connectivity
to remote network
 Trunks placed in
Route Groups
 Route Groups placed in
Route Lists
 Use Route Lists to cycle
through alternate trunks
 Use retry timers and counters
to move through alternate
trunks quickly
 Use return code mapping (SIP
codes to Q.850) to stop
advancement
Redundancy and Load Balancing — Multiple Trunks
Administration->Call Routing->Route/Hunt-
>Route Pattern
Route Pattern contains Route List
Administration->Call Routing->Route/Hunt->Route
List
Route List contains Route Groups
Administration->Call Routing->Route/Hunt->Route
Group
Route Group contains individual
trunks
Redundancy and Load Balancing — Multiple Routes
 Create redundant routes for

Route
the same destination
Pattern
A trunk denotes a unique route
Route  Trunks are placed in Route

First List Second
Groups
Choice Choice A distribution algorithm in
Route Route Route Group determines
Group Group which trunk will be used next
First
Choice
Second
Choice
Third
Choice  Route Groups are placed in
Route List
IP WAN SBC
IP PSTN or  A Route Pattern contains the
PSTN Route List
Redundancy and Load Balancing — Route
Advancement
 Adjust the SIP Timers

for optimal behavior
Lower ―Retry Count for
SIP INVITE‖
Works for both TCP
and UDP connections
 Adjust re-routing parameter

to stop re-rerouting on
certain cause codes
Q.850 cause code
obtained from reason
header or derived from
SIP return code
Redundancy and Load Balancing — Using DNS
_sip._udp.cucm IN SRV 10 1 5060 cucm1.cucm.com

IN SRV 10 2 5060 cucm2.cucm.com
DNS Server cucm1.cucm.com IN A 50.50.50.5
cucm2.cucm.com IN A 50.50.50.6
 Defining multiple trunks using

INVITE INVITE: 50.50.50.5 DNS hostname
No Reply  Contrast DNS with the IP
CUCM1 address configuration
Next entry in DNS SRV
Additional DNS lookup step,
done for every call, may cause
INVITE: 50.50.50.6
some extra delay
180 Ringing A highly available DNS server
180
200 200 OK is required
BYE CUCM2 Advancement in DNS stopped
BYE
if any answer received
200 200 OK
Redundancy and Load Balancing — Using DNS
_sip._udp.cucm IN SRV 10 1 5060 cucm1.cucm.com

CUCM Cluster IN SRV 10 2 5060 cucm2.cucm.com
DNS Server IN SRV 10 3 5060 cucm3.cucm.com
Device Pool NY IN SRV 10 4 5060 cucm4.cucm.com
cucm1
SIP Trunk from Device Pool NY PSTN

cucm2
Device Pool NJ  Both phones can make

outbound calls to PSTN
cucm3  Inbound calls will work as long
as the calls are sent to one of
cucm4 the subscribers in Device
Pool NY
Comparing DNS and IP Address Defined Trunks
Parameter DNS Defined Static IP Defined
Configuration Centralized in the DNS Server Need to be updated when

necessary
Redundancy Automatic Multiple Trunks must be defined
(Multiple Destinations) manually
Load Balancing Automatic Must be defined into Route
(Between Multiple Trunks) Groups and Route Lists
Network Resources Must have highly available DNS No DNS Server required
Server
Network Traffic DNS Lookups to resolve address No extra traffic needed
adds network traffic
PBXs/
Enterprises
Manager 8.0(1)
Gateways

Manager 8.0(1)
SAF-enabled
Conferencing
Cisco Unified
Presence
Unified
Messaging
Microsoft
LCS/OCS
Cisco Unified SCCP
IBM
Personal Lotus MGCP
Media Resources
 MTP
Early Offer with choice of
codecs
INVITE w/ SDP Either G.711 or G.729

(Offer: Ph IP/Port) INVITE w/ SDP DTMF conversion
(Offer: MTP IP/Port)
In-band  Out Of Band
180 Ringing 180 Ringing
(RFC-2833 & KPML/Notify)
200 OK w/ SDP
(Answer)
 Transcoder
200 OK w/ SDP Codec Mismatch
(Answer: MTP
MTP IP/Port) ACK
 Trusted Relay Points
ACK
Network Services such as
Media Media QoS marking
Usage of Media Resources
Media
 MRGL contains MRGs
Step 1: Choose the
Resource
Highest MRG with an
Available Device of the Manager  MRGs are read in order as
Type Required they were configured
Step 2: Round Robin (Mrg1, Mrg2, Mrg3, etc.)
Load-Balance Between Media
Devices of the Same Type
Within an MRG
Resource  Unified CM will allocate
Group List
1st 2nd resources in this order: Mrg1
Choice Choice first, if resource in Mrg1 is
exhausted then Mrg2, if
Media Media resource in Mrg2 is exhausted
Resource Resource
Group 1 Group 2 then Mrg3
1st 2nd
Choice Choice  The resource in each MRG
Media Media Media
is round robin based on the
Resource Resource Resource most available capacity of
1 2 3 each device
Media Resources Comparison
Unified CM Built-in SW ISR HW ISR
Service
MTP – Early Offer G.711a or G.711u G.729, G.729a, G.729b, (Configured as Transcoder)
G.729ab, G.711a, or G.729, G.729a, G.729b,
G.711u, iSAC, iLBC G.729ab, G.711a, or
G.711u, iSAC, iLBC
MTP – DTMF Relay Yes Yes Yes
Conference Bridge (Audio) G.711a, G.711u, and Supported on Can support mixed mode
Wideband Meetingplace Express conferences
Media Server
Conference Bridge (Video) None Supported on Trans-rating, Higher
Meetingplace Express Quality
Media Server
Music On Hold Codecs: G.711a, G.711u, Supported with or without No Support
G.729a. Unicast or SRST function. Multicast
Multicast
Annunciator G.711a, G.711u, G.729a No Support No Support
Trusted Relay Points No Support Yes No Support
Transcoding No Support No Support G.711 to G.729, G.722,

iLBC, GSM, etc.
RSVP Agent No Support Supported with MTP and Supported with MTP and
Transcoders Transcoders
Technology Basics — Security
 Done through SIP Security

Profile associated with trunk
 Device Security Mode set to
Encrypted for AES-128 crypto
string
 Transport Type set to TLS for
signaling security
 Digest Authentication for SIP
message level challenge
 X.509 Subject Name set to
correspond to peer name
 Media Encryption (SRTP) done if
transport type is TLS and both
sides offer compatible capabilities
Technology Basics — Security
 Certificate import using Platform Administration

Certificates may be obtained directly from peer entities (self
signed) or from a certificate authority
Ensure that the X.509 name matches that on the certificate
 TLS Sessions – May be shared for multiple calls

 Security Interworking with UCME, CUBE, and Gateways
New in 8.0
Service Types
IME
SAF/CCD
EMCC
End to End RSVP
Session Management Edition Deployment Model
New in 8.0 - SIP Trunk Service Types
 New field Trunk Service Type

Sets the trunk up for specialized functions
New in 8.0 - Inter-company Media Engine Enterprise
C
1. IME servers obtain DID patterns from their

CUCM and join IME network
2. IME Servers publish DIDs to IME network IME Servers
3. First Call is placed over PSTN

4. Both CUCMs send Voice Call Records
(VCRs) to their IME Servers Public
5. The Originating IME Server verifies call Internet
with the Terminating IME Server
6. The Originating IME Server pushes Enterprise
Enterprise
learned SIP route to that DID to the A B
originating CUCM ASA
7. The Originating CUCM saves the route
8. When another call is placed to the same
DID, the call now goes over the Internet
as an IME call via the learned IME SIP
route w/ UC capability PSTN
New in 8.0 - IME Trunks
 Dynamically Created Connections within one single trunk definition

 No pre-configured destination address; otherwise just like ordinary
SIP Trunks
 Just one configured virtual trunk is sufficient for all IME connections
 Configuration Tasks
Set Retry count for SIP Trunks to 2 (quick failover to PSTN)
Enable the trunk to accept Out-of-Dialog REFER (for failover to PSTN)
Define appropriate transformation/translation should be defined to handle
E.164 DNs (to handle calls that failover to the PSTN)
Set security profile should be non-secure (ASA will establish TLS and
convert to SRTP)
 Changes in SIP Signaling
Proprietary headers between CUCM and ASA (Ticket for verification,
Headers for PSTN fallback)
ASA measures QoS and may initiate a REFER to CUCM for failover to
PSTN
The Unified CM SIP Trunk BRKUCC-2403
Understanding the
New in 8.0 - IME Trunks Intercompany Media Engine

Solution
SIP Trunks for Inter-company Media Engine
Trunk Usage Placing calls over the Internet once the destination DID and address are verified
Destination Addressing IPv4 or DNS (A or SRV)
Destination Redundancy Multiple destination addresses may be provided. More than one connection may
be used under the control of a ―virtual‖ Route List
Source Redundancy CMGroup associated with the configured trunk’s device pool provides multiple
source servers
Failover to PSTN If QoS measured by ASA is worse than configured; if SIP connection can not be
made
Security Encrypted calls over the Internet
DTMF As configured and negotiated
Services Voice and Video
Capacity (Max number of No pre-set limit

concurrent connections)
Configuration Specifics OOD REFER and SIP retry timer
New in 8.0 - Service Advertisement Framework/Call Control
Discovery
San Jose Cluster New York Cluster
Advertises Pattern: Advertises Pattern:
8408XXXX/4:+1408576 8212XXXX/4:+1212496
Advertises Address: 10.1.1.1 Advertises Address: 10.5.1.1
Learns Pattern: Learns Pattern:
8212XXXX/4:+1212496 SAF Forwarder 8408XXXX/4:+1408576
Learns Address: 10.5.1.1 Learns Address: 10.1.1.1
SAF-enabled
IP Network
SAF Client (External) SAF Client (External)
8408XXXX
Publishes on AS1 8212XXXX
Publishes on AS1
IP : 10.1.1.1-3 SIP IP : 10.5.5.1 - 3 SIP
Pattern:<p d="4:+1408555"> Pattern:<p d="4:+1212555">
8408XXXX</p>
10.1.1.1 PSTN 8212XXXX</p>
10.5.1.1
San Jose Cluster
New York Cluster
SAF: The underlying infrastructure for capability information distribution

CCD: A service that uses SAF for distributing call routing information
New in 8.0 - SAF Trunks
 Similar to IME trunks but intra enterprise and not

inter enterprise
 Dynamically created – do not have any pre-
configured destination address
 One trunk hosts several individual connections
New in 8.0 – SAF Trunks for CCD
SIP Trunks for Call Control Discovery
Trunk Usage Placing calls over the IP Network once the destination is determined using Service
Advertisement Framework
Destination Addressing IPv4 and IPv6 only – no DNS A or SRV records
Destination Redundancy Multiple destination addresses may be provided. More than one connection may
be used under the control of a ―virtual‖ Route List
source servers
Failover to PSTN Routes come with PSTN prefixes that allows re-routing if necessary
Security Can not be defined on SIP Trunks that use authentication and/or encryption
DTMF As configured and negotiated
Capacity (Max number of No specific limit other than system limits

Configuration Specifics No specific configuration requirements
New in 8.0 - Extension Mobility Cross Cluster
User’s EM profile
Visiting TFTP Home TFTP
Visiting CM DB Home CM DB
7 change
notify
Hmm… Modify
Do you have
user not 6 device
local
record userid?
4 Lookup User
2
Yes
3
Visiting CCMCIP Visiting EM 5 Home EM
Visiting CM cluster Receive Home CM cluster
Home TFTP
Address
1
Login using
Reset userid, PIN 9
Receive
8
Configuration Phone’s config in visiting. Original
Phone’s config in visiting. Tainted (a.k.a. mini-config)
User’s EM profile config in home
Visiting phone EMCC base phone in home
Phone’s logged in config in home

New in 8.0 - EMCC Trunks
 Service specific trunks separate from ICTs connecting

different clusters
 One trunk configuration sufficient for all EMCC connections
 Does not communicate EMCC login information (that is done
through the EM service)
 Calls normally completed through the home cluster
 Allows E911 and PSTN calls to be handled from visiting
cluster (using local GWs) if Local Route Groups are
configured
 Allows media resources (currently RSVP agents) to be
allocated from visiting cluster rather than the home cluster
 Defines some proprietary headers to instruct visiting cluster to
complete calls to PSTN
Extension Mobility –
New in 8.0 - EMCC Trunks Cross Cluster
SIP Trunks for Extension Mobility Cross Cluster
Trunk Usage For PSTN (including E911 calls) only
Destination Addressing IPv4 and IPv6
Destination Redundancy Primary and backup addresses are communicated via the EMCC service
source servers
PSTN Access If local route groups are configured, then PSTN calls are routed through the
visiting cluster so as not to send media over the WAN
Security As configured
DTMF As negotiated
Services All line side services
Capacity (Max number of No preset limit

Media Resources RSVP agents if required are allocated from the visiting cluster. However other
media resources may still be allocated from the home cluster
The Unified CM SIP Trunk – New for 8.0
New in 8.0 - E2E RSVP Trunks
 Extends QoS by allowing the establishment of

RSVP with devices outside of the Unified CM
cluster
 Reduces the number of media resources (RSVP
Agents) per call when establishing calls to devices
beyond the Unified CM cluster
 It is a standards-based End to End CAC
mechanism
The Unified CM SIP Trunk – New for 8.0
New in 8.0 - Local RSVP vs. End to End RSVP
SIP Trunk
WAN
Local QoS – RSVP between Local QoS – RSVP between
local RSVP agents local RSVP agents
Local QoS – No preconditions
SIP Trunk
WAN
Media
SCCP End to End QoS – RSVP between local and remote RSVP agent
SIP End to End using preconditions
RSVP
New in 8.0 - End to End RSVP - Basic Call Flow
 Initial INVITE contains
preconditions:
INVITE a=curr:qos e2e none
a=des:qos mandatory e2e
183 Session Progress sendrecv
PRACK
 Recipient responds
200 OK (PRACK) similarly before
RSVP Agent RSVP Agent forwarding INVITE:
PATH a=curr:qos e2e none
RESV a=des:qos mandatory e2e
sendrecv
PATH a=conf:qos e2e recv
RESV
 RSVP Agents are
UPDATE allocated and reservation
200 OK (UPDATE) performed
180 RINGING Reservation state
200 OK (INVITE) indicated in UPDATE/200
ACK (INVITE) OK
INVITE (w/o SDP)  SDP exchanged after
200 OK (INVITE) w/ SDP reservations confirmed
ACK (INVITE) w/ SDP  Media negotiation
continues as usual
Advanced RSVP Design
New for 8.0 - E2E RSVP Trunks and Deployment in UC 8.0

Solutions
End 2 End RSVP SIP Trunks
Trunk Usage For calls with bandwidth reservation for media
Destination Addressing Same as ordinary Trunks
Destination Redundancy Same as ordinary Trunks
Source Redundancy Same as ordinary Trunks
Failover Configuration dependent (Mandatory or Optional)
Security SRTP will be negotiated during media set up time (not during reservation time)
DTMF Same as ordinary trunks
Capacity (Max number of Same as ordinary Trunks

Configuration Specifics Enable E2E RSVP in the Device Configuration page associated with the trunk
Session Management Edition
IP
PSTN
CUBE CUBE
Unified CM
Session
Manager
Cluster
Leaf Unified CM H323 Annex M1

Clusters/ Leaf MGCP
UC Systems SIP
Cluster 1 Cluster n QSIG PBX QSIG PBX Q931 PBX Q931 PBX
• SME - A new deployment model for UCM Clusters with a central

cluster providing aggregation for multiple leaf call control agents
Session Management Edition (Release 7.1(3))
 Dial Plan Aggregation with Centralized Routing for

Multiple UC Systems
 Centralized TDM/IP PSTN connections
 Centralized Applications
 Integration or Migration of Existing 3rd Party PBXs
Although, an SME deployment has little to do with SIP Trunks specifically, the working of
the whole SME concept depends on how well the trunking model works – particularly with
regard to the feature set extended across multiple trunk hops, interoperability with other
vendor’s IP-PBXs, and of course, the usual considerations for scalability, load-balancing,
and redundancy.
There are no new developments for the SIP Trunk for this deployment model nor is there a
particular service type defined for the trunk as in some other cases. There are also no code
differences between leaf cluster CUCM and a session manager CUCM.
The Unified CM SIP Trunk - Session Management Edition
SIP Pre-Conditions for RSVP over SIP Trunks
SME can route SIP Preconditions, so that end to end RSVP reservations can be
created for calls from Leaf Cluster to Leaf Cluster, Leaf cluster to IOS Gateway, or
Gateway to Gateway
SME
RSVP over SIP Trunks
Central
Central Site 2
Site 1
IP WAN
RSVP Agent RSVP Agent
SCCP/SIP SCCP/SIP
Media
Stream
Phone 1 RSVP RSVP Phone 2
Agent Agent
RSVP
Branch 1 Reservation Branch 2
SAF Design
IP PSTN DN Ranges and ―To PSTN‖ prefixes
advertised by SME only – Leaf
clusters listen but do not advertise
CUBE CUBE
SME
PSTN PSTN PSTN
SFO Dallas NYC
SAF and SME can be combined today –

+ SME can use SAF to distribute Internal DN ranges and To PSTN Prefixes
All intercluster IP calls route via SME
If SME is unreachable – Leaf cluster route calls to the local PSTN
If Leaf cluster is unreachable – SME routes calls to PSTN
- Some of the dynamic nature of SAF is lost – Leaf cluster state availability
External Call Control – Policy Server
PSTN
IP PSTN
CUBE CUBE
4) Setup (cgpn=19725550100,
cdpn=14695550101)
Policy
SME 3) XACMLRes(permit,continue, Server
modify callingnumber=+19725550100)
2) XACMLReq (mcgpn=+19725550141,
mcdpn=+19725550101)
SME Administrator
Leaf UC
assigns an
Systems External Call
CUCM ADMIN
Control Profile to a
translation pattern
CUCM
Cluster 1
CUCM
Cluster n
PBX m
1) Dial 914695550101
to intercept the
outbound call and
apply policy
Bob
Enterprise A +19725550141
Inter-Company Media Engine (IME) B2B Communications
PSTN
IP PSTN
Authenticate
CUBE CUBE
IME ASA IME Trunk ASA IME
SME
CUCM
Cluster
Enterprise B
Leaf UC
Systems Internet
PSTN Gateway
ASA IME
CUCM CUCM PBX m
Cluster 1 Cluster n
CUCM
Cluster
Enterprise A Enterprise X
Session Management Edition Trunk Characteristics
Session Manager SIP Trunks
Trunk Usage Traditional usage – to other Unified CM Clusters, Other IP-PBX, Gateways for
calls
Destination Addressing IPv4, IPv6, or DNS (A or SRV)
Destination Redundancy Several trunks to different addresses organized in Route Groups and serviced by
Route List
source servers
Failover to PSTN Governed by Automatic Alternate Routing
Security Full TLS and SRTP is supported
DTMF Any of Out-of-Band and In-Band methods supported
Services Voice, Video, and Fax
Capacity (Max number of No pre-set limit

Configuration Specifics No special configuration required
Unified CM SIP Trunk
Gateway, CUBE and CUSP Integration
Types of Gateways
Comparison of Gateway Protocols
Benefits CUBE brings to Unified CM Native SIP Trunk
Usage of CUSP with CUBE
PBXs/
Enterprises
Manager 8.0(1)
Gateways

Manager 8.0(1)
SAF-enabled
Conferencing
Cisco Unified
Presence
Unified
Messaging
Microsoft
LCS/OCS
Cisco Unified SCCP
IBM
Personal Lotus MGCP
Unified CM SIP Trunk — Gateway Integration
Technology Basics
TDM IP  Types of Gateways

MGCP, H.323, and SIP
 Deployment
PSTN
PRI Layer 3
Layer 2 Q.931 Backhaul over TCP
Framing MGCP over UDP
Central site
MGCP Gateway Cisco Unified
CM Distributed at branch locations
 Selection
PSTN
PRI Layer 3
Layer 2
Framing
SIP over UDP/TCP/TLS Route Partitions to select
particular Route Patterns
SIP Gateway Cisco Unified
CM Route Patterns to route calls to
GWs via Route Lists and
Groups
PSTN
PRI Layer 3
Layer 2 H.225
Framing H.245
H.323 Gateway Cisco Unified
CM
Unified CM SIP Trunk — Gateway Integration
Gateway Features by Protocol
H.323 SIP MGCP
Centralized Provisioning
QSIG Tunneling No, but some services
are supported
Centralized CDR (DS0 Granularity in Unified
CM CDR)
MLPP (Preemption)
Hook-flash Transfer with Unified CM

ISDN Overlap Sending No GK Legend:
ISDN FAC IE Name Display
Yes
NFAS
No
AT&T Megacom NSF Partial Partial
With Caveats
SRTP (Unified CM to GW)
Mobility Manager VXML-Based Voice Profile
Mgmt
Interstar Xmedius T.38 Fax Server
Caller ID on FXO
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
TDM T3 Trunks
ISDN Video Switching on GW
Set numbering Plan Type of Outgoing Calls
G.Clear (Clear Channel Data) Support
H.320 Video
Unified CM SIP Trunk – Gateway Integration
Recommendations
Application H.323 MGCP SIP Preferred
Voice Mail Control of Individual MGCP

(3rd Party) Ports
Configuration Dial Peer Based Centralized in CUCM Dial Peer Based MGCP
Load on CUCM Least MGCP
Q.SIG Only between PBXs Supported MGCP

Tunneling
Video (IP/VC) H.320 ISDN H.320 ISDN H.323/SIP
Fax and Modem Pass-through, T.38 Pass-through, T.38 Pass-through, T.38 H.323/SIP/
MGCP
Port Density High Density Cards MGCP
Redundancy Range of options with Range of options H.323/SIP

dial-peers with dial-peers
Security IPSec and SRTP IPSec and SRTP TLS and SRTP SIP
Voice XML Supported Supported H.323/SIP
Unified CM SIP Trunk — Gateway Integration BRKUCC-2010
SIP Gateways Deployment Considerations Designing UC Gateways and

DSP Engineering in
Enterprise
Networks
Parameter SIP Digital Gateways
Early Offer/Delayed Offer Support both Early and Delayed Offers
Codec Support G.711, G.729, G.722, iLBC
Media Resource Requirement Support both delayed and early offer without MTP
Redundancy & Load Balancing From CUCM, can define multiple Gateways. From Gateway,
can define multiple CUCM servers with load balancing and
redundancy
Authentication and Security Digest Authentication, TLS, SRTP
DTMF Method RFC-2833, SIP-NOTIFY, KPML
Special Configurations None Required
Cisco Unified Border Element
Key Features CUBE
Session Mgmt Demarcation

Real-time session Mgmt Fault isolation
Call Admissions Control Topology Hiding
Ensuring QoS Network Borders
PSTN GW Fallback L5/L7 Protocol Demarc
Statistics and Billing Statistics and Billing
Redundancy/Scalability
Interworking Yours Security

H.323 and SIP Encryption
SIP Normalization Authentication
DTMF Interworking Mine Registration
Transcoding SIP Protection
Codec Filtering FW Placement
Fax/Modem Support Toll fraud
Unified CM SIP Trunk – Cisco Unified Border Element
Integration
CUBE Usage Within The Enterprise
SP VoIP
 As a protocol converter
SIP SBC between Unified CM 4.x and
H.323 Unified CM 5.x and later
CUBE versions
SIP
Unified CM 4.x
Unified CM 5.x +
CUBE
H.323 SIP
SIP SIP  As a interconnection point
between Unified CM, other IP-
PBX, and SIP Application
Servers
SIP App
Unified CM 4.x Unified CM 5.x+ IP-PBX Server
Unified CM SIP Trunk — Cisco Unified Border Element
Integration
CUBE Basics and Usage in SIP Calls
 CUBE services to enhance

Unified CM Native Trunks
voice service voip
H.323 to SIP Translation
allow-connections sip to sip Delayed to Early Offer
allow-connections h323 to sip
sip
early-offer forced
No MTP, Multiple Codecs
Codec Filtering
Cisco Unified DTMF Relay
Border Element Transcoding
SIP Message Normalization
H.323/SIP SBC SIP Pro-active Connection
Trunk Trunk Monitoring
Integration
CUBE Redundancy and Load Balancing
 Device Pool defined with list of
subscribers in CM Group
 SIP Trunk with the device pool
defined to each CUBE
CUBE  SIP Trunk from each CUBE

defined to each server in CM
Group
CUBE
 RG on CUCM cluster with each
trunk
 Dial-peer hunt usage on CUBE
CUBE
May use ICMP Ping or SIP
Options
Router(config)#dial-peer hunt ?
<0-7> Dial-peer hunting choices, listed in hunting order within each choice:
0 - Longest match in phone number, explicit preference, random selection.
1 - Longest match in phone number, explicit preference, least recent use.
2 - Explicit preference, longest match in phone number, random selection.
Dialpeer hunt— 3 - Explicit preference, longest match in phone number, least recent use.
group for selecting 4 - Least recent use, longest match in phone number, explicit preference.
5 - Least recent use, explicit preference, longest match in phone number.
dial-peers 6 - Random selection.
7 - Least recent use.
Integration
CUBE Usage in Delayed to Early Offer
 Some SIP servers may

require SDP Offer in
initial INVITE (Early Offer)
INVITE w/o SDP
INVITE w/ SDP  By itself UC Manager
(Offer: CUBE IP/Port) requires MTP resources for
180 Ringing Early Offer
200 OK w/ SDP
(Answer)  CUBE can convert a
CUBE
Delayed Offer INVITE from
200 OK w/ SDP Unified CM to Early Offer
(Offer)
ACK INVITE w/o MTP
ACK w/ SDP
(Answer)  CUBE manages offer-
Media Media answer negotiation on
both sides
Integration
CUBE Usage in Multiple Codecs Offerings
voice service voip

 Unified CM with MTP can
allow-connections sip to sip only offer one of either
allow-connections h323 to sip
sip G.711 or G.729 codecs in
early-offer forced
Early Offer
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
 CUBE may offer several
voice class codec 2 codecs in outgoing INVITE
codec preference 1 g711alaw
on behalf of UC Manager
dial-peer voice 100 voip
destination-pattern 901144T
voice-class codec 2  Codecs may be ordered as
dial-peer voice 101 voip
desired to suit preferences
destination-pattern 91T
voice-class codec 1
Unified CM SIP Trunk—Cisco Unified Border Element
Integration
CUBE Usage Codec Filtering
Incoming Dial-Peer Outgoing Dial-Peer

dial-peer voice 919 voip dial-peer voice 9190 voip
incoming called-number 919.... destination-pattern 919....
Filter
session target sipv2 session target sipv2
Cisco Unified
codec transparent codec g729r8
Border Element
g729r8, g711ulaw g729r8
Proposed Terminating
Codecs: Incoming Outgoing
Codec:
g729r8, g711ulaw, Call Leg Call Leg g729r8
g7xx-non-std
 Tailor offered codec to requirements

Fax pass-through may require G.711
Branch phones may require G.729
Integration
CUBE Usage Trunk Monitoring
A
SP CUBE
CUBE
CUBE
Options Ping Options Ping Options Ping
SP fails over to CUSP marks CUBE as CUBE marks CUCM

―secondary IP address‖ down and reroutes calls as down and rejects
for trunk and/or to alternate CUBEs in incoming calls from
reroutes incoming calls the stack SP
Options Ping Options Ping Options Ping
CUSP rejects incoming CUBE shuts down dial- Not yet supported
calls peer to CUSP and
rejects incoming calls
from CUCM
 CUBE monitors both legs

Temporarily shuts unresponsive dial-peers
Results in more reliable and faster call completion
Unified CM SIP Trunk — Cisco Unified SIP Proxy Integration
CUSP Basics
 CUSP helps manage

large SIP based networks
Integrated in ISR – no need
for separate server
Dial Plan
SIP Trunk Aggregation
Load Balancing and
Availability
Normalization and Flexible
Routing
Unified CM SIP Trunk — Cisco Unified SIP Proxy
Integration BRKUCC-2305
CUBE Scaling: 2,000-15,000 Sessions SIP Trunk Integrating Voice and Video
over IP Networks Using the
Cisco Unified Border Element
CUBE ISR
SP SIP
CUCM SIP Trunk CUCM
SP SIP Trunk
CUBE A
CUBE
SBC
CUBE
CUBE
CUBE Cluster
CUBE + CUSP
SP SIP
Trunk CUCM SIP
SP SIP Trunk
CUCM
SBC A
CUBE
CUBE
CUBE
CUBE CUBE Cluster
CUBE ASR
SP SIP
CUCM
SP SIP Trunk CUCM SIP Trunk
A
SBC
CUBE (Enterprise)
Usage in Intercluster Trunks
Describe Intercluster Trunks
ICT Connection Comparison
Connectivity options with H.323 and SIP
PBXs/
Enterprises
Manager 8.0(1)
Gateways

Manager 8.0(1)
SAF-enabled
Conferencing
Cisco Unified
Presence
Unified
Messaging
Microsoft
LCS/OCS
Cisco Unified SCCP
IBM
Personal Lotus MGCP
Unified CM SIP Inter Cluster Trunks
H.323 ICT Services
 ICT based on H.323

 Q.SIG based services
Call completion on busy
Call completion on no-reply
Path replacement
Allows PBX interconnection and
migration
 Voice media encryption
 Call preservation
 Wide codec selection in
faststart
 Call Admission Control
Based on static locations
 Integration with UCME
using CUBE
H.323 ICT — GK Controlled
 Can either have a full mesh

network or one that is
controlled by a Gatekeeper
 Gatekeeper
Provides dialplan
management and call
admission control between
SBC clusters
SBC GK
GK Provides shared access to
common resources
Integration with UCME using
via-zone CUBE (Optional) for
maintaining H.450 based
supplementary services
H.323 ICT (GK) — Call Flow
CMGroup Members CMGroup Members
(attached to Device Pool for ICT) (attached to Device Pool for ICT)
ARQ ARQ
Node1, Node2, Node3 Node4, Node5, Node6
ACF ACF
H.225
H.245
Cluster 1 Cluster 2
 When a cluster registers with GK, all nodes in the CM group automatically
register with GK
GK knows status of C1-Node1, Node2, Node3 & C2-Node4, Node5 & Node6
 If any Node goes down, it gets reflected in GK registered GW list

 Call signaling goes between Cluster Nodes and GK provides reachability
and other functions (CAC) for the CUCM which acts as GW
Towards SIP ICT
 ICT based on SIP
 Can either have a full mesh
network or one that is
controlled by a SIP Proxy
 Cisco Unified SIP Proxy
provides dialplan
management, message
SBC
routing, alternate routing and
SBC load balancing
ISR
ISR
 Call Admission Control done
via static locations
 CUBE not required for UCME
Integration for supplementary
services as UC Manager
supports REFER
Unified CM SIP Inter Cluster Trunk
Full Mesh SIP ICT
Device Pool
Device Pool
SIP Trunk Needs to be per Host (Or Use DNS-SRV)

 TCP based SIP trunk provides faster failover
 Adjust retry configurations for INVITEs from default to 2 on both
Unified CM and CUSP, for faster failover
 Call setup goes between Cluster Nodes and media between
end-points
 Device-Pool Assignment activates SIP demon in CUCM Nodes
Unified CM SIP Inter Cluster Trunk
SIP ICT With CUSP
Device Pool
Device Pool
 CUSP acts as a aggregation point (like GK in H.323)

 CUSP supports Options-Ping, to verify the SIP daemon status of
CUCM nodes
 Call signaling & media negotiation does not flow directly between
individual subscribers
 CM Group Nodes listed in the Device-Pool that assigned to the SIP
trunk, activate SIP daemon in those Nodes only
SIP Trunks vs. H.323 Trunks
H.323 H.323 w/Q.SIG SIP
Annex M1 Features / Q.SIG Tunneling
Signal Authentication
Media Encryption
GK Support
SIP Proxy Support
iLBC and G.Clear Support
G.722 Support
Multicast MoH
SIP Subscribe/Notify, Publish –
Presence
Path Replacement
Call Completion to Busy Subscriber
Call Completion No Reply
Message Waiting Indicator (On /Off)
Alerting Name
Legend: Yes No Not Applicable

H.323 GK vs. SIP CUSP
H.323 GK SIP CUSP
Global Dial Plan
Call Admission Control Via Zone Bandwidth Via Location
Settings Settings
Signal Encryption Via IPSec Via TLS
Hierarchical Structure for Scalability
Multiple Trunks for Redundancy and Load Balancing
Clustering for Redundancy
Endpoint Tracking Through Through
Registration Monitoring
Number Translation With External
Server via GKTMP
Time of Day Routing With External
Server via GKTMP
DNS Lookup for Destination With External
Server via GKTMP
Normalization of Messages
Legend: Yes No With Caveats

PBXs/
Enterprises
Manager 8.0(1)
Gateways

Manager 8.0(1)
SAF-enabled
Conferencing
Cisco Unified
Presence
Unified
Messaging
Microsoft
LCS/OCS
Cisco Unified SCCP
IBM
Personal Lotus MGCP
Unified CM Express - SIP vs. H.323 Integration
H.323 SIP
Supplementary Services Automatic detection of Unified CM Does not automatically detect calls
calls – Does not use H.450 for and will use SIP REFER or 302 for
supplementary services and call re-direction; will resort to
instead hairpins call signaling hairpin if these fail
Unified Border Element CUBE can inter-work between CUBE is not necessary for SIP
H.450 based UCME and ECS integration. UCME can either use
based CUCM; can also avoid REFER or ReInvite
media hair-pining.
If using GK based clustering, can

use via-zone construct to insert
CUBE in the call path
Services Supplementary services such as Call features plus Presence
forward, transfer, conference, etc. information exchange
DTMF H.245 based RFC-2833, SIP-NOTIFY, and KPML
Video Basic video calls only (SCCP Not supported

phones only) – no supplementary
services
Unified CM Express - SIP Trunk Integration
Considerations
Parameter Unified CM Express
Early Offer or Delayed Offer Supports both Early and Delayed Offer calls from the Unified
CM (But generates Early Offer calls only)
Call Features Unlike Unified CM (which only hairpins), Unified CM Express

may use both forms of redirection – REFER/302 (only if
dialplans are compatible); if not, use hairpin
Other Services Presence is supported; video is not
Authentication, Signaling and Media Authentication and TLS are supported; SRTP is not
Encryption
Redundancy and Load balancing Unified CM Express cluster can be used for load-balancing
and redundancy
Configuration Requirements Configure SIP Trunk to accept Replaces header and REFER
messages in Unified CM
Configure allow connections sip to sip in Unified CM Express
Usage in Voice Messaging Integration
Integration Considerations with
Unity
Unity Connection
Usage of Cisco Unified SIP Proxy for Efficiency and Scalability
PBXs/
Enterprises
Manager 8.0(1)
Gateways

Manager 8.0(1)
SAF-enabled
Conferencing
Cisco Unified
Presence
Unified
Messaging
Microsoft
LCS/OCS
Cisco Unified SCCP
IBM
Personal Lotus MGCP
Unified CM SIP Trunk – VM Integration
Integration Considerations
 Message Waiting Indication (MWI) Delivery

(Unsolicited Notify Only)
 Type of Divert, for calling/called party ID routing info to Unity/Unity
Connection
(Diversion-header, RPID1)
 Supported Codecs
(G.711, G.729, iLBC1, G.7221)
 Transfer method from Unity/Unity Connection to IP PBX
(REFER)
1Supported by Unity Connection only
Unified CM SIP Trunk – Unity Connection
Comparing SIP and SCCP Integration Methods
Parameter SCCP SIP Integration

Integration
Ease of configuration More steps Few steps
Failover Supported Supported
MWI DN configuration Required Not required
TSP requirement Not needed Not needed
Feature Parity Same as SIP Same as SCCP
TLS / SRTP Support Supported Supported (UC 7.x)
Unified CM SIP Trunk – Unity
Comparing SIP and SCCP Integration Methods
Parameter SCCP Integration SIP Integration

Ease of configuration More steps Few steps
Failover Supported Not Supported
MWI DN configuration Required Not required
TSP requirement Needed Not needed
Feature Parity Same as SIP Same as SCCP
TLS / SRTP Support Supported Not Supported
Unified CM SIP Trunk – Cisco Messaging Integration
Comparing SIP Integration with Unity and Unity Connection
Parameter Unity Unity Connection
Codec Support G.711u-law, and G.729 G.711a-law, G.711u-law,

G.729, iLBC, and G.722
Re-routing info Diversion-Header Diversion-Header & RPID
MWI Unsolicited Notify Only Unsolicited Notify Only
Transfer Method REFER REFER
TLS/SRTP Not Supported Supported
Failover Not Supported Supported
MTP* Not needed Not needed
DTMF-Relay RFC 2833 RFC 2833
*Older SCCP phone models may require a Media Termination Point (MTP) to function correctly for DTMF-relay.
Unified CM SIP Trunk – Cisco Messaging Integration
Direct (Without CUSP)
Ports (1-24)
Serial
T1 Ports(25-48)
 Multiple SIP trunk connections with different PBXs

 Ports are pre-allocated for each of these PBXs
With CUSP
Ports (1-48)
Serial
T1
 Unity Connection has single integration to CUSP

PIMG
 PBXs SIP trunk integration with CUSP
 CUSP allows sharing of Unity/Unity Connection ports with
PBXs
Advantage – higher utilization of available Unity Connection Resources
Using CUSP with multiple VM servers
Connection
Unity
CUSP
ISR
Serial TIMG
T1
Unity
Serial TIMG
T1
PIMG
CUSP Reroutes the traffic based on Source IP address of PIMG/TIMG

CUSP is aggregating multiple Unity/Unity Connections
Conferencing – MeetingPlace and Video
H.323 and SIP Integrations
Unified CM as a protocol converter
Video MCU integrations
As a conferencing resource
Direct with SIP Trunk
PBXs/
Enterprises
Manager 8.0(1)
Gateways

Manager 8.0(1)
SAF-enabled
Conferencing
Cisco Unified
Presence
Unified
Messaging
Microsoft
LCS/OCS
Cisco Unified SCCP
IBM
Personal Lotus MGCP
Unified CM SIP Trunk – MeetingPlace Integration
CUCM and MeetingPlace 8.0
SCCP for Ad-Hoc

Conferences
Scheduling MeetingPlace
Application SIP Trunk for Application MeetingPlace
Server Web Server
Scheduled
Conferences
SCCP MP
conf Express
bridge Media
Server Scheduling
Application
MP HW
Outlook Conf. Meeting
Video Scheduling Bridge Scheduling
Audio Video Audio Access

Audio Recording
Internal Ad-hoc Video Audio
Video
conference Users Audio/Video
conference
Internal MP User user Internal MP User
Unified CM SIP Trunk – MeetingPlace Integration
SIP Integration Method
SIP Trunk
CUVC-MCU
CUVC-MCU
 Type of transport protocol – UDP or TCP (TCP Recommended)

 Failover - Yes; multiple CUCM nodes supported
 DTMF supported - RFC2833, KPML, In-band tones
 Signaling authentication - No
 Signaling encryption - No
 Media encryption - No
 Codec Supported - G.711 / G.722 / G.729 / iLBC
 MTP Required - No (enabling MTP blocks video)
 Media Resources - Choose one without MoH resources to avoid disruptions
 Any SIP profile tweaking - No
Unified CM SIP Trunk – Video
Technology Basics
Third-Party Third-Party H.323,

SCCP/SIP SIP Video
Video Endpoints
Endpoints
Unified Personal Unified

Communicator Communications
Manager
Cisco CUVC
GK H.320 Gateways
Unified Video
IOS H.323
Advantage 7985 Gatekeeper
MTP
Cisco CUVC
H.323, SIP
MTP, Transcoders, Conference
RSVP Agents Bridges
with Pass-Through + MeetingPlace
Codec Support CUVC SCCP 9971/9951 Integration
Conference Bridges
Traditional Video Conferencing Using H.323
Small
Branch
Office
QoS
Medium PSTN
Enabled
Branch Office ISDN
IP WAN
 Consists of one or more Video Infrastructure

Gatekeeper zones Zone1
Gatekeeper
Gateways
 H.323 Video Endpoints,
Gateways, and MCU register with GK
GK MCUs
the Gatekeeper
 Gatekeeper performs dial plan
resolution and bandwidth H.323
management Video
terminals
Headquarters
SIP Usage in Video Telephony
 SCCP and SIP Endpoints
Registered with CUCM
 H.323 Endpoints statically
configured in CUCM
RTP
SCCP
H.323 CUVC-MCU
SIP (Ad-hoc Audio & Video Bridge)
Integration based on H.323 or SIP Trunks
Unified CM H.323
SIP Proxy Gatekeeper
SIP H.323
Trunk Trunk
SIP H.323
MCU
Ad-hoc SCCP SIP
Conf
SIP VC Video Telephony H.323 VC

Endpoints Endpoints Endpoints
H.323 and SIP Video Features
Video Features H.323 SIP
Video Codec Support H.261, H.263, H.263+, H.264 H.261, H.263, H.263+, H.264
Authentication and Security Use IP sec TLS
Video escalation down to audio Supported Supported
Video Escalation from Audio Call needs to be Video to start, Supported
then can be De-escalated to Voice
and Escalated to Video
Usage of Media Resources Must not have ―MTP Required‖ Must not have ―MTP Required‖
checked (RSVP agent with Pass- checked (RSVP agent with Pass-
through config or TRP can be an through config or TRP can be an
option) option)
Call Administration Locations based through CUCM Locations based through CUCM
Endpoints registration External gatekeeper needed Supports endpoint registering

MCU interworking (SCCP is an Supported Supported
additional integration available)
Telepresence Endpoints No support Supported
Telepresence Integration Using CUVC-MCU or MXE Using CUVC-MCU or MXE
MOC / SameTime Integration Not Supported Supported
Dual Media for Presentation Supported (H.239) with Ver7.1X Not Supported
fhr8CX83
For More Information: http://www.facebook.com/CiscoCollab
Q&A
vbharga@cisco.com
Key Takeaways
 SIP is gaining popularity and SIP implementation in

Cisco Unified Communications product portfolio is
expanding rapidly
 A single protocol to inter-work all devices provides
seamless integration and a richer user experience
 SIP Trunk on the Cisco Unified Communications
Manager is the key to make this integration happen.
It is not just for PSTN access
 Understanding the capabilities and working of this
SIP Trunk is essential to building and deploying a
Cisco Unified Communications network
 SIP may not yet be the best protocol for all possible
integrations
Related Sessions
 BRKUCC-2006: SIP Trunk Design and Deployment

 BRKUCC-2010: Designing UC Gateways and DSP Engineering in
Enterprise Networks
 BRKUCC-2012: Understanding SIP Endpoints in Cisco Unified
Communications Manager
 BRKUCC-2040: Extension Mobility – Cross Cluster
 BRKVVT-2305: Interconnecting Voice and Video over IP Networks
Using the Cisco Unified Border Element
 BRKUCC-2403: Understanding the Intercompany Media Engine
Solution
 BRKUCC-2931: Case Study of a Large Scale Centralized SIP
Trunk Implementation
 BRKUCC-3099: Advanced RSVP Design and Deployment in
Unified Communications 8.0 Solutions
Recommended Reading
 Cisco Unified Communications

Solution Reference Network
Design (SRND) for Cisco
Unified Communications
Manager Release 8.x, available
online at:
www.cisco.com/go/designzone
 SIP Trunking: Migrating from
TDM to IP for Business to
Business Communications by
Hattingh, Sladden, and Swapan
(ISBN: 1-58705-944-4)
Complete Your Online
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for each session evaluation
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SIP Trunk Design and Deployment

Playbook for the Enterprise

Cisco Unified Communications

 Unified CM SIP Line Side

SIP Line Interface provides SIP

 Unified CM SIP Trunk has lots

 Use the SIP Trunk

 Use the SIP Profile

Redundancy, Availability and Load Balancing

Media Resource Requirements

 Why does the Unified CM

 What about media clipping?

 Video Bit-rate between and within regions

Unified CM default is 384,000 bps (CIF, 30 fps)

 SIP Trunk is defined on one

Administration->System->Cisco Unified CM Group

 Create redundant routes for

Route  Trunks are placed in Route

 Adjust the SIP Timers

 Adjust re-routing parameter

_sip._udp.cucm IN SRV 10 1 5060 cucm1.cucm.com

 Defining multiple trunks using

_sip._udp.cucm IN SRV 10 1 5060 cucm1.cucm.com

SIP Trunk from Device Pool NY PSTN

Device Pool NJ  Both phones can make

Parameter DNS Defined Static IP Defined

Configuration Centralized in the DNS Server Need to be updated when

Cisco Unified Communications

INVITE w/ SDP Either G.711 or G.729

Trusted Relay Points No Support Yes No Support

Transcoding No Support No Support G.711 to G.729, G.722,

 Done through SIP Security

 Certificate import using Platform Administration

 TLS Sessions – May be shared for multiple calls

End to End RSVP

Session Management Edition Deployment Model

 New field Trunk Service Type

1. IME servers obtain DID patterns from their

3. First Call is placed over PSTN

 Dynamically Created Connections within one single trunk definition

New in 8.0 - IME Trunks Intercompany Media Engine

SIP Trunks for Inter-company Media Engine

Destination Addressing IPv4 or DNS (A or SRV)

DTMF As configured and negotiated

Services Voice and Video

Capacity (Max number of No pre-set limit

SAF: The underlying infrastructure for capability information distribution

 Similar to IME trunks but intra enterprise and not

DTMF As configured and negotiated

Services Voice and Video

Capacity (Max number of No specific limit other than system limits

Phone’s config in visiting. Tainted (a.k.a. mini-config)

User’s EM profile config in home

Visiting phone EMCC base phone in home

Phone’s logged in config in home

 Service specific trunks separate from ICTs connecting

New in 8.0 - EMCC Trunks Cross Cluster

SIP Trunks for Extension Mobility Cross Cluster

Trunk Usage For PSTN (including E911 calls) only

Destination Addressing IPv4 and IPv6

Services All line side services

Capacity (Max number of No preset limit

 Extends QoS by allowing the establishment of