Sip Training

SIP
Training
2008 Avaya Inc. All rights reserved. 1

Introduction to SIP
Session Initiation Protocol
SIP is an application layer signaling protocol for
establishing, modifying, and terminating multimedia
sessions including audio, video, chat. Extensions also
support instant messaging and presence.
Is a text-based request/response protocol modeled after
HTTP.
Uses URLs for addressing (sip:, tel:, mailto: ).
SIP is a very simple protocol with very few client/server style messages
What is Session Initiation Protocol (SIP)?
http://www.ietf.org/html.charters/sip-charter.html
SIP is an IETF application layer-protocol that can establish, modify, and terminate
multimedia sessions
RFC 3261
Media agnostic
Voice, video, instant messaging, etc.
Media negotiation
Offer-Answer model
Similar to HTTP
Request-Response model
Text message-based protocol
Easy to debug
Reuses other IETF protocols
UDP, TCP, TLS, DHCP, DNS, SDP, RTP, MIME, etc.

Components of SIP
User Agent
User Agent Client
Generates and sends SIP requests and receives responses
User Agent Server
Receives SIP requests and generates SIP responses
Registrar
Provides mapping of logical SIP addresses to physical SIP addresses
Location Service
Used by SIP Proxy or Redirect server to obtain the mapping from logical
SIP addresses to physical SIP addresses
Proxy Server
Forwards SIP requests downstream and responses upstream
Redirect Server
Generates 3xx responses directing clients to contact an alternate set of
URIs
Presence Server
Acts as a Presence Agent or proxy server for SUBSCRIBE requests
Avaya incorporates all these functions in the
SIP Enablement Services
Basic SIP Functionality & Terms
User Location: locating the device to be used for a particular session.
Presence: in the SIP world, this is the status of the principal and is open or
closed, or some other mutually exclusive value.
User Availability: determining the willingness of a party to engage in
communication
User Capabilities: determining supported media types and parameters to be
used
Session Setup: ringing, establishment of caller and callee session parameters
Session Once the media offer/answer exchange is completed we have a
session. Each session is generally associated with a single dialog.
Session Management: modifying, transferring or terminating sessions.
Transaction consists of a single request and any response to that request.
Invite + Ack is a transaction.
Dialog Peer to Peer between 2 UA

What is SIMPLE?
http://www.ietf.org/html.charters/sipping-charter.html
SIP for Instant Messaging and Presence Leveraging

Extensions
IETF working group
Introduces Presence into communications state
Builds on RFC 3265
Now a standard: RFC 3856
Selected as basis for 3gpp networks & applications

What is SIPPING?
http://www.ietf.org/html.charters/sipping-charter.html
Session Initiation Protocol Project INvestiGation

IETF working group
Chartered to document the use of SIP for several
applications related to telephony and multimedia
SIPPING-19 refers to SIP Services Examples draft
draft-ietf-sipping-service-examples-08
19 example telephony features implemented in SIP
Purpose is to ensure that basic features interoperate
Other SIPPING items
SIP Basic Call Flow Examples (RFC 3665)
Message Waiting Indication (RFC 3842)

Number Portability Nirvana
The SIP Address of Record (AOR)
SIP provides a single user identity, the public address
e.g. sip:tomcain@avaya.com or sip:7204448464@avaya.com
User identity maps to any number of devices
Hoteling and User Mobility are native to SIP
SIP Phone
Softphone
Instant Messaging PDA Dual-Mode

Mobile Phone
Traditional phone
SIP Messages
Requests (Methods) Responses
1xx: Provisional
REGISTER
request received, continuing to
Register contact information process the request
INVITE, ACK 2xx: Success
Setting up sessions the action was successfully received,
understood, and accepted
BYE, CANCEL
3xx: Redirection
Terminating sessions further action needs to be taken in
OPTIONS order to complete the request
Querying servers about their 4xx: Client Error
capabilities the request contains bad syntax or
cannot be fulfilled at this server
SUBSCRIBE, NOTIFY (RFC 3265) 5xx: Server Error
Event notification framework the server failed to fulfill an apparently
MESSAGE (RFC 3428) valid request
Instant messages 6xx: Global Failure
the request cannot be fulfilled at any
server

Other Common Messages
183 Session Progress
More information for the call that may not be classified.
PRACK
Provisional ACK for reliability.
487 Request Terminated
No session but a transaction.
Refer
To transfer a call usually.
Notify
Valuable information for the subscriber.

RTP/Media
Inband/Out of Band
System Parameters IP Options
Payload type
Trunk Form
Set type
Many vendors use different SIP stacks.
CM load
Programming on 3.1 is different from 5.0.
SES load
Very confusing

SDP
RFC 2327
Session Name and Purpose
Time the session is active
Information to receive the media on.
Specifies Audio/Video Profiles and correlates them to a
number used in SDP and RTP.
Allows dynamic assignment of bandwidth and
unassigned types.
http://www.iana.org/assignments/rtp-parameters

SDP
RFC 2327 Message Body

Major Difference between SIP and H.323
SIP can be used for more than telephony and video
SIP is text-based; H.323 is binary
Dial tone and ring-back tone is generated locally by the
phone
Dial plan resides in the phone
Digits are not transmitted to the PBX until the phone
completes digit collection based on dial plan
SIP clients can be represented by alpha-numeric ids and
passwords
SIP users can associate multiple devices with a single AoR

SIP Identification
UserID tomcain@avaya.com
Handles tomcain@avaya.com; 48464@avaya.com;
7204448464@avaya.com
Handles get resolved to a contact
Contact sip:48464@135.8.70.200 (station at CM)
Multiple Contacts make up ContactSet
ContactSet is assigned to a USERID

Avaya SES Terminology
SES (Sip Enablement Services) Lots of info available on the Enterprise Portal
under Products -> Sip Enablement Services
The SES can be comprised of 3 different configurations of servers
An Edge knows about all users and which Home the users register to speaks
to all Homes and an outbound proxy has Master Administrator privileges
Only 1 Edge per Domain
The Home is where the client is registered with speaks to the Edge, other
Homes through the Edge, and CM
Up to 20 Homes per Edge
A Combo has the functionality of both a Home and an Edge
Only 1 Combo in a solution
Same hardware and software for all during setup, tell each server its function
and only those services are enabled to change requires a re-install

Edge and Homes
company.com
Mapping INVITE sip:joe@company.com

Edge Data
Edge proxy
INVITE sip:joe@company.com
Home Home Home

proxy proxy proxy
Homes
User User User

Data, Data, Data,
Prefs. Prefs.
Prefs.
company.com company.com company.com

Interop
SIP brings multi-vendor interoperability
Open standard which allows for a mix and match of endpoints and
communication with other vendors proxies
Supported configurations are based on completed tested by the Solutions and
Interoperability Lab and posted application notes

How does SIP work?
SIP uses a series of Request and Response messages
INVITE, CANCEL, BYE, REFER, ACK, etc
1xx, 2xx, 3xx, 4xx, 5xx, and 6xx
Uses RTP/RTCP for audio

SIP Request Messages
INVITE - Indicates that the user or service is being invited to participate in a
session
ACK - Confirms that the client has received a final response to an INVITE
request
BYE - Indicates that the user wishes to terminate the call
REGISTER - Indicates that a User Agent is attempting to add its address to the
Redirect server's user database
CANCEL - Cancels a pending request but does not affect a completed request
OPTIONS - Queries the capability of the servers
MESSAGE send an instant message
SUBSCRIBE/NOTIFY SIP events

SIP Response Messages
1xx Information Responses
For example: 180, Ringing
2xx Successful Responses
For example: 200, OK
3xx Redirection Responses
For example: 302, Moved Temporarily
4xx Request Failures Responses
For example: 403, Forbidden
5xx Server Failure Responses
For example: 504, Gateway Time-out
6xx Global Failure Responses
For example: 600, Busy Everywhere

Example Call Flow
Alice Proxy Bob

INVITE sip:bob@example.com
407 Proxy Authentication Required
ACK sip:bob@example.com
100 Trying Bob
180 Ringing answers
180 Ringing
200 Ok
200 Ok
Bob
RTP hangs up
BYE sip:alice@example.com
BYE sip:alice@example.com
200 Ok
200 Ok

Invite
User A 5060 User B 5061
Request URI
Header
Body

Invite
Indicates that a user or service is being invited to
participate in a session. The body of this session
message would include a description of the
session to which the callee is being invited.
The contact identifies to the callee where it came
from.
CSeq changes with each transaction.

100 Trying

180 Ringing

200 OK
The contact will identify to the caller what the address of the callee is.
The request has succeeded.
The information returned with the response depends on the method used in the request.

ACK Confirms that the client has received a final response to an invite request, and is only used with an invite request.

BYE

200 OK

SIP Endpoints
ECP Training

Endpoint Credentials
Seems most SIP endpoints
have the same credentials
in some form or another.
User Name
Login/password
Registrar
Domain
Proxy

Endpoints
Depending on the endpoint greatly depends on
what functions you wish to utilize.
Some businesses have SIP call center phones
Soft Clients
Hard Phones
IM clients
IVR apps
B2BUAs

Avayas Endpoints
46xx SIP
HTTP/TFTP
Text file required
Speed Dial functions only
96xx OneX Deskphone
HTTP Only
Text file required
PPM aligns CM Station form with SIP endpoint.
OneX Desktop
Softclient
OneX Mobile Dual Mode
16xx SIP
Call center Endpoint (Avaya Proprietary)
LAB
Work on the endpoints
Understand the files HTTP server
SIP vs. H.323
Option 242 vs. 176
SES
Add users
Add stations
Use clients
Tracing
Endpoints
Interop notes
http://www.avaya.com/gcm/master-usa/en-us/produ
cts/offers/sip_enablement_services.htm&View
=ProdResources#Application%20Notes
Support Endpoints
https://enterpriseportal.avaya.com/ptlWeb/internal/
products/P0276/Interoperability#

SIP Trunking and Service
Providers
ECP Training

This Session
Understand Devices For the purposes of this
SBC discussion we will have a
Border Elements CM, SES, SBC and
Infrastructure Service provider in the
topology unless otherwise
Call Flow
stated.
Routing
Design
Know your topology
Troubleshooting
The right questions to ask

What is SIP Trunking?
Its the ability to provide trunk service from a Carrier using IP instead of
traditional ISDN.
Less hardware
For both the customer and the service provider
Reduced cost
In many cases up to 65% less
Reduction in floor space
Gives back 80%
Faster implementation
Operational Savings

SIP Trunking and Mobility
Session Border Controller

SIP VoIP
SIP Service Provider
Signaling
Enterprise
VoIP
Media Internet
Residential
Firewall
SIP
Registration SIP Phone
WiFi SIP
& Signaling IM
Enterprise
SIP Enabled
Firewall
CM Features
Remote Enterprise Employees
Connected via Internet w/o VPN

SIP is Control SIP
SP
SBC
ISDN or
ISDN
G860

Border Elements SBCs
Where are they used?
A Session Border Controller is a
device used in some VoIP
Acme Packet networks to exert control over
the signaling and usually also
the media streams involved in
setting up, conducting, and
Covergence
tearing down calls.
A typical use is when NAT
traversal is performed at a
InGate firewall. A standard firewall can
not change a SIP packet to
reflect the right IP address but
an SBC can.
Voip Engine

What are ALGs/SBCs?
http://tools.ietf.org/internet-drafts/draft-wing-session-auth-00
.txt
Two techniques commonly provide ingress/egress security today --
firewall application layer gateways (ALG) and Session Border
Controllers (SBC) [5]. In order to perform its security function, the ALG
must be able to examine the SIP signaling, whereas an SBC is actively
involved with modifying the SIP signaling (specifically the SDP, and
often SIP headers as well). However the use of hop-by-hop signaling
encryption (such as with TLS) breaks firewall ALGs, and the use of
end-to-end signaling encryption (such as S/MIME) or end-to-end
integrity protection (such as SIP Identity [3]) breaks SBCs. Additionally
both a firewall and an SBC constrain the media path, causing packets
to not take the best path. A firewall requires the signaling and media
flow through the firewall (which is achieved through network design)
and an SBC causes media to flow through the SBC (by modifying SDP
in SIP signaling).

Controllability
The SBC has less hardware
because the PSTN programming
is controlled by the Service
provider.
SBC
VS
The G860 management is on
both the T3 side and the CM
side.
The difference is that the SBC

receives the trunking as SIP and
the G860 converts the trunks to
SIP from ISDN.
G860

D-Marks
Its important to know the topology
D-Marks vary depending on equipment in place and whats owned by
the customer.
For example there are situations where the SP will deliver and
maintain the SBC for the customer. In this scenario the
demarcation is the last Avaya owned device before this equipment.
This could be an SES or a CM if an SES is not installed.
Its also very difficult to know what the devices are w/o proper notes
on the equipment involved. Sending a trace w/o the IPs and
equipment names makes it harder for the technician helping w/ the
trouble.

Network Information
Information Needed
Border Elements
Names of devices
Acme Packet, Nextone, etc.
Loads
Not on just 1 product but several
Equipment involved
CM, SES, Nortel, etc.
Endpoints
Avaya, Cisco, what?
What does the customer expect this solution to do?
There are many false expectations as the new technologies
emerge. Some are assumed others are sold that way.

Be Aware NAT/Firewall
Communication Mgr
IP & SIP
NAT
On-net
SES 36xx PIX

Home/Edge Legacy 38xx 515E
Cisco Proxy
PSTN Off-net
GW
PSTN
SIP
Avaya SIP 3rd Party SIP
Services Services
NAT traversal
Network Address Translation
Its basically when the address presented on the inside of the firewall is
changed to present a different address on the outside of the firewall and vice
versa.
Whats this got to do with SIP?
In a regular firewall the traversal only manipulates the IP in the header of the
message at the physical layer. With SIP technology if the body of the
message doesnt reflect that same change then the packets are directed to
the wrong place. Usually they are directed to the firewall instead of the
device inside/outside the firewall hence they get dropped.
Terms you may hear
Sip ready, enabled
This means it will accept SIP packets but in most cases will not change the packet
at the application layer.
Results of a poor firewall installation?
Calls drop after 32 seconds.
One way Voice path.
Works inbound but not outbound.
Wheres the Voip?
CO MPACT
N
CO MPACT
Y/N
PSTN
Y N
C OM PACT
SES
Y Y
H.323 SIP
P HONE /E X IT P AGE P AGE OP T IO NS
LEF T RIG HT
PHONE /EXI T S PE AK ER HO L D
P AG E PAGE OP TI ONS
L EF T R IGHT
SP EA KE R HO LD HE AD S ET TRAN S FER
ABC D EF
1 2 3
M U TE C ON FE RE NCE
HEADS ET TR ANS FE R
GHI J KL M NO
ABC DEF
4 5 6 DROP
M UTE
1 2 3 CONF ERENCE
PQ RS T UV W XYZ
7 8 9
RE DIAL
GHI JKL MNO
4 5 6 DROP
PQRS
7
TUV
8
WX YZ
9
RE DIAL * 0 #
* 0 #

Where am I going?
S87xx SES
SIP Trunking
Modular Messaging
Diamond Meeting Exchange
PSTN
Voice Portal

Rules
Call flow

Call Flow
A local SES Endpoint call uses the COR of the defined station in CM
to place a call.
Call Still traverses CM even if its a SIP to SIP call
An ISDN call may use Proxy Route Select.
This is on the locations form in CM.
When using a Proxy Select Route pattern only the trunk matters.
This field is used for any URI based messages that dont know
where to go in CM.

Calls within the same SIP domain
SES will:
Lookup to see if the user placing the call is local
Checks to see if the user has an extension defined. If it does
then it goes to the CM defined extension for processing via the
assigned clan to that extension.
The proxy then routes the call to the registered contact for that
user.
If the user placing the call is not local it looks for MAPS
Address maps should not conflict!
Maps essentially punch a hole in your security
Refer to the SES Install and Admin Guide Chapter 2
If the user is not local it will then turn to the Edge server for info on
where the user resides.

Call Flow Rules
Foreign Domain
If the call is to a foreign domain SES will:
Use the outbound proxy on the Host if it has been
configured.
If no outbound proxy, attempt to resolve the address
in DNS lookup.
If no outbound proxy or DNS found then it will send it
to the Edge which will attempt the same process
before ultimately sending a reject of no route found.

Outbound
Know your topology

Who am I?
Where do I point to?
There are probably many ways to do this but the most common
are:
(Most Common) If the Managing SBC is the SPs SBC then
thats where it will point.
If the SBC is managing the routing of the calls then the
customer may ask for it to be pointed to the Inside SBC. Either
way its a decision of the Network Engineer.
Remember some customer networks may not need an SBC
because they are on a network that requires no NAT traversals
such as an MPLS network.
What about AAR and ARS routing.
In the case of no SES CM AAR/ARS routing rules apply as they
normally have.

Call Flow #1 OB
SBC
CO MPACT
5
CO MPACT
4
3
PSTN
CLAN
MAP Firewall
2
C OM PACT
SES 1
M UTE
PQRS
7
SP EA KE R
HEADS ET
GHI
4
1
TUV
8
PHONE /EXI T
JKL
5
ABC
2
WX YZ
9
P AG E
L EF T
6
MNO
3
DEF
RE DIAL
PAGE
R IGHT
DROP
OP TI ONS
TR ANS FE R
CONF ERENCE
HO LD
*
M U TE
PQ RS
7
4
0
SIP
S PE AK ER
HE AD S ET
GHI
1
T UV
8
P HONE /E X IT
J KL
#
ABC
W XYZ
9
P AGE
LEF T
M NO
6
3
D EF
RE DIAL
P AGE
RIG HT
DROP
OP T IO NS
TRAN S FER
C ON FE RE NCE
HO L D
* 0 #

Call Flow #2 OB
CO MPACT
CO MPACT
2 3
PSTN
1 C OM PACT

LEF T RIG HT
L EF T R IGHT
ABC D EF
1 2 3
GHI J KL M NO
ABC DEF
4 5 6 DROP
M UTE
1 2 3 CONF ERENCE
PQ RS T UV W XYZ
7 8 9
RE DIAL
GHI JKL MNO
4 5 6 DROP
PQRS
7
TUV
8
WX YZ
9
RE DIAL * 0 #
* 0 #

Common questions to ask
You will find that many questions asked on ISDN are asked for SIP
trunking.
Do I send a 1?
Are you sending me 10 or 11 digits?
Is there a DID range?
Do I need to send a + for International?
How many digits do I send for International?
Are these trunks both inbound and outbound?
Do we route pattern choices if this fails.
Backup plan

Where does this lead?
Address maps are a feature in the SES used to route SIP messages to
the appropriate SIP hosts based on the destination address in the
INVITE message.
These are typically for devices that are not registered.
Maps are an important part of the call flow process.
Maps are very important in being able to troubleshoot where the
breakdown may be happening.
Trusted hosts are for devices that you dont want the SES to issue a
challenge and response to.

SES Programming OB
The Host Map is best practice
for outbound calls to the SSP.
This can be done from the
home but can be confusing
for others checking for
errors.
All rules start with ^sip:
Brackets signify range of that
digit.
Curly Brackets signify replication
to include the previous
command.
The left window states any 10
digit number beginning with 2-9
may be considered.

SES Programming OB
The Map is the rule as the
Contact is the route for that rule.
When you add a contact to a
Host it is not automatic as with a
media server.
The Host does not have a
definition by default where a
media server does.
Media servers are assigned
to clans when built.

SES Programming OB
$(user) gets replaced with
the user of the original
request.
The IP Of SBC can be ours
or theirs depending on how
its managed.
Transport is typically UDP
but there have been
installs as TCP so be sure
to ask.

SES Programming OB
Notice that there is both a
rule and a route in this
picture.
There can be multiple
rules for one route.
There can be multiple
routes for one rule.

Inbound
Know your topology

Call Flow #1 IB
SBC
CO MPACT
CO MPACT
2 1
3 PSTN
CLAN
MAP Firewall
4
C OM PACT
SES
5
SIP
LEF T RIG HT
L EF T R IGHT
ABC D EF
1 2 3
GHI J KL M NO
ABC DEF
4 5 6 DROP
M UTE
1 2 3 CONF ERENCE
PQ RS T UV W XYZ
7 8 9
RE DIAL
GHI JKL MNO
4 5 6 DROP
PQRS
7
TUV
8
WX YZ
9
RE DIAL * 0 #
* 0 #

Call Flow #2 IB
CO MPACT
CO MPACT
2 1
PSTN
3 C OM PACT

LEF T RIG HT
L EF T R IGHT
ABC D EF
1 2 3
GHI J KL M NO
ABC DEF
4 5 6 DROP
M UTE
1 2 3 CONF ERENCE
PQ RS T UV W XYZ
7 8 9
RE DIAL
GHI JKL MNO
4 5 6 DROP
PQRS
7
TUV
8
WX YZ
9
RE DIAL * 0 #
* 0 #

Programming Trunks
Where do they point to?
Distributed or not all 3rd party traffic points to the
Edge as best practice.
If there is no SES then it will look at the CLAN.
This is referred to as SIP Direct trunking.
Minimum load of 4.0.1 of CM.

SES Programming IB
Media Server Maps
Best practice is to use the Media Server maps for
passing calls to CM.
The SP is not a registered device to the SES, only
a trusted device. So calls that hit the SES need to
know where to go.
Same principal in CM for inbound calls, if they are
not programmed into AAR/ARS routing then they
fail.

SES Programming IB
Contacts are automatically generated because when you

select a map on a Media Server it already knows the IP of
the CLAN.

Basic Call Flow Registered endpoint
S87xx
SES
Near End CLAN!
PSTN

Troubleshooting
CID 121397
1st step is to see if there is traffic coming in and
what that traffic is.
Easily done tracing the call:
From CLI in the Edge server
ccstrace on file=/var/home/ftp/pub/out1
Tail f ~ftp/pub/out1
ccstrace off
Wait for traffic to hit and watch to see if the packet is
correct.
Troubleshooting
Tethereal trace
A great trace for individual endpoints is as follows:
tethereal -R 'sip contains "30261@avaya.com"
Wireshark
tethereal -R sip -w /tmp/trace.pcap
Live Verbose
Tethereal R sip V
<ctrl> C should stop the trace.

MST Trace

MST without toolsa - Requires Root
1: Setup the MST in CM selecting both "Log MST" & "Trace Analyzer".
The file will be saved to /var/log/ecs as a .log file. This should be the latest
file to date.
su - sroot
2: You then have to type "logmst" + filename and redirect it to a folder.
ex. logmst 2005-1024-33433.log > /tmp/output
If you have multiple files with saved MST then a >> can be used to the same destination destination.
logmst 2005-1024-3345.log > /tmp/output
################################################################
### The above commands extract the MST messages from the log ###
################################################################
### The below command converts it from hex to text ###
"mst" 26 lines, 915 characters########################
3. mta -s guts -s sum /tmp/output > ~ftp/pub/output.trace

Intelligent Communications for Any Office
SIP Trunking to IP Service
Providers
SIP through SES to
network Quick Edition
offices to other Avaya
systems
Global analog and BRI
SIP
gateways
LAN
Centralized bulk provisioning /

SW upgrades across multiple
Quick Edition offices
Select key system features
Teleworker functionality

G860 Overview
ECP Training

Overview
Up to 6000 protected channels in a small
footprint (5U) chassis
High Density Interfaces DS3, OC3,
STM-1
Lower Cost of Ownership
Reduction in Carrier Access Charges
Lower costs associated with acquiring and maintaining data center
floor space
Operational and maintenance cost savings by reducing hardware
footprint (less power, UPS, cooling needs)
SIP Connectivity to Avaya Communication Manager
Carrier Grade Reliability
Works in Conjunction with G650s
Ideal for Large IP-based Contact Centers and Campuses
Installed and Supported by Avaya Global Services

Target Market
Need for at least one DS3 (28 T1s, 672 channels)
The higher the number of Port Networks required to support T1s,
the more cost effective it becomes to replace the T1s with T3s in a
G860
Large Contact Centers
~300+ agents
Large Campuses
~1,000+ phones
Mostly IP environment (H.323 or SIP stations)
But also supported in digital, analog and hybrid environments

Chassis

EMS
EMS Client controlled.

EMS Client

Ethernet Card
Port 20

Shelf Controller
Run solaris
These have individual addresses as well as a virtual for
the active.
These are what push the configuration to the TP cards.

TP Card
Trunking in from provider.

In this lab its coming from an Adtran that is muxing 6
PRIs.
The $$ cards, the powerhouse of the cabinet.
In/outbound trunking is from the primary IP.
Lab Flow
SIP

G860 R2 Chassis Configuration
Slots 7 to 10 are used for up to 4 TP-6310 boards (including the
redundant TP-6310 board) according to the customers requirements:
3 Protected DS3 (1+1)
1 Protected STM1/OC3 (1+1)
Each OC-3 PSTN connection is a cage provided with a slim form
pluggable SFP 155 Mbps optical module to connect to an optical fiber
with an STM-1/LC-type optical connector
Each T3 PSTN interface port is an SMB connector with Tx and Rx
transceivers

Solution View Each TP-6310
(3DS3 DSP
module) is
controlled by its
corresponding
CM Server
Business Center A
Dual provider
redundancy
strategy
IP
PSTN CM - A
G860 - A
See Figure 1A for
additional details
CM - B
IP
Business Center B
G860 - B

Possibilities
G860
ISDN WAN
Covergence
(SBC)
Home2
CM
CoreRouter
DO
Home1

Technical Features
Standard 19, 5U packaging
All configurations come with critical availability
1+1 shelf controllers
1+1 Ethernet switches
1+1 alarm functionality
2+1 load-sharing power supplies
4+1 redundant fans
1+1 redundant power inlets
Redundant backplane (2x 17Gbps)
NEBS III compliant (DC version)
Hot swappable modules
Optional protected trunk interface
3 x 1DS3 +1DS3
3 x 3DS3 + 3DS3
3+1 OC3
3+1 STM1
Element Management System (EMS)

SES Introduction and Best
Practices
ECP Training

SES Ever Changing
First load 2.0 was completely manual for
installation.
Last release combined SES and CM on the same
backplane and shared the same Linux files.
Know what releases you will be dealing with.
Understand the complete configuration.
Multiple CMs to a single Home not multiple
Homes to a single CM.

SES
Lock Eth0 to 100F
Use Edge for SSPs.
Use Homes for CM trunking.
Plan Maps out ahead of time to prevent conflicts.

Endpoints
Manage with 46xxsettings.txt files when possible.
46xx use TFTP/HTTP but the 96xx only use HTTP.
Most other SIP endpoints use a .txt file in some
manner.
PPM failures are typically the result of SES not
able to log into CM.

2007
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