This chapter provides a high-level overview of the interworking
between the Multimedia Communications Platform (MCP) and the
following systems:
•Primary Rate Interface (PRI)-enabled switches
•SIP-T-based switches
•third-party g ateways
•traditional phones
•third-party vo icemail servers
Interworking with PRI-enabled switches
The MCP uses the SI P PRI Gateway to perf orm interworking with
PRI-enabled switches (switches with PRI interfaces).
For more detailed information about the SIP PRI Gateway, please refer
to the MCP SIP PRI Gateway Basics document.
3
Functional description
The SIP PRI Gateway is a signaling and media gateway that
interconnects a SIP-based Voice over IP (VoIP) domain and a system
that uses ISDN Primary Rate Interface (PRI/Q.931).
A PRI-enabled system can be a switch that works on the
Public-Switched Telephone Network (PSTN) or Private Branch
Exchange (PBX). The Nortel Networks DMS-100 and the Nortel
Networks Meridian SL-100 are examples of PRI-enabled switches.
The SIP PRI Gateway’s primary functio n is to convert the packet-based
voice streams of the V oI P system to circuit-based vo ice streams of the
PRI-enabled system.
The SIP PRI Gateway supports the following call types:
•SIP to PRI
•PRI to SIP
•PRI to PRI
The SIP PRI Gateway supports the following call features:
•Basic call: The ability to make a call.
•Hold/Retrieve: The ability to hold and retrieve calls.
•Call transfers: The ability to forward the call to a third party after the
call is established. Call transfer is limited to SIP clients. Callers from
the PRI-enabled switches cannot perform this function.
•Call redirect: The ability to forward a call before it is answered.
•Codec negotiation: The ability to negotiate between different VoIP
codecs during call setup, mid-call, call transfer, and call retrieve.
The SIP PRI Gateway supports the following codecs:
•Call rejection: The ability to reject a call on nodal authentication
request.
•Calling party name and number: The MCP supports the
delivery/reception of calling party name and number information
to/from PRI-enabled switches. Incoming callin g party informa tion
privacy indication is honored by the MCP . For MCP call originations,
privacy indication is not used.
•Dual-Tone Multi Frequency (DTMF) outpulsing: The ability to
outpulse to PRI-enabled sw itches. However , DTMF detection is not
provided. Military DTMF digits A-D are current l y not supported.
•ISDN trunk group selection: The selection is based on information
provided in a request Universal Resource Identifier (URI).
•Ringback: The SIP PRI Gateway provides ringback towards the
circuit-switched side of the network.
•PRI variant support: The SIP PRI Gateway supports different PRI
variants. For a list of the PRI variants that the SIP PRI Gateway
supports, please refer to the MCP SIP PRI Gateway Basics
document.
•SIP/PRI mapping support, including:
— Protocol parameter mappin g
— error codes
— PRI cause values with SIP responses
— presentation a nd screening indicators
•PRACK: the provisional response acknowledge message that
ensures that the ringing signal does not get lost.
•T ype of Server (ToS): The ToS bit can be configured to indicate the
priority of the voice packet over data packet to ensure quality of
service.
•Long-call service: The ability to detect abandoned calls.
More detailed information about the functional capabilities of the SIP
PRI Gateway is available in the MCP SIP PRI Gateway Basics
document.
Interworking with CS 2000
Even though Nort el Ne tw orks C om m unication Server 2000 (CS 2000)
is a V oIP switch, it d oes not fully suppor t the Session Initiati on Protoco l
(SIP). The following de scri p ti on provides information on how the MCP
interworks with the CS 2000.
Overview 5
The MCP does not support VoIP over Asynchronous Transfer Mode
(ATM) or pure ATM, as the MCP network nodes are IP-based and not
ATM based.
Functional description
The interworking betw een an MCP and CS 2000 uses Session Initiat ion
Protocol for Telephones (SIP-T) over User Datagram Protocol (U DP) to
transport ISDN User Part (ISUP), the call control part of the Signaling
System 7 (SS7) protocol.
SIP-T is an e xtension of the Se ssion Initiation P rotocol (SIP) that allows
SIP to be used to facilitate the interconnection of the Public Switched
Telephone Network (PSTN) with p acket networ ks. SIP-T e ncapsu lates
the ISDN User Part (ISUP) messages in the SIP messages and
translates ISUP information into the SIP header for routing purposes.
Although the MCP SIP Application Module supports receiving of
encapsulated ISUP messaging (usi ng SIP-T), it does not send
encapsulated ISUP back out. For more information about the
capabilities of the SIP Application Module, please refer to the SIP Application Mo du le Basi cs document.
Figure 1 shows a network view of the MCP/CS 2000 interconnection.
The MCP configuration enables direct connection between the MCP
and CS 2000 with no intervening SIP proxies. The CS 2000 and its
associated media gateways are placed in the private managed
network.
Figure 1 MCP/CS 2000 Network view
network 1
network 1
Supported services
This section provides inform ation on services that can interwork
between the two platforms .
PSTN
SIP-T
MCP
SIP-T
SIP
SIP-T
PSTN
network 2
SIPSIP
MCP
network 2
VPN Dialing
A Virtual Pr ivate Network (VPN) d ial plan enables an enterprise to have
one common dial plan across different geographic locations without
incurring long-distance expenses.
The MCP system uses domains to man age its call routing, while the CS
2000 system uses customer groups.
A profile is used to facilitate call routing between the two systems. The
header named “x -nor tel- pro file” is used to ident ify the CS 20 00 pro file.
The profile is used to ma p the domain to a PSTN custome r group in the
PRODOMAIN table in the Database Module.
The profile is added when a new do main is created in the MCP system.
Note: The profile name in the TEL EPROF table on the CS 2000 side
must match the profile name in the PRODOMAIN table on the MCP
side. Otherwise, interworking between the MCP and CS 2000 will not
work.
Call Forward
Both the MCP and the CS 2000 support call forwarding. No extra
interworking consider ati ons are necessa ry as call forw ardi ng does not
interact across the MCP and CS 200 0 domains, except to deliver a call
between the two domains.
Call Transfer
Call transfer works between the two platforms. The calls are transferred
because from each platfor ms view the other is ju st ini tiatin g a new call.
Ad Hoc Conference calls
There is no change in the conference application between the two
platforms. Whe n an ad hoc conference call is made, the client
application that starts the conference controls where the conference
bridge is allocated. For exa m ple, i f the conference call is made from a
client application on the MCP side, the conference bridge will be
allocated on the MCP.
Overview 7
Media Negotiation
The MCP provides media negotiation for the CS 2000 since the CS
2000 gateway contr ol ler i s not capable of providing this function in the
current release.
The MCP requires a list of all commonly supported codecs across the
CS 2000 media gateways to be provisioned in the cs2k.xml file. The list
is required because the MCP does not determine which gateway the
CS 2000 will use.
Although the MCP also allo ws f or the pr ovision ing o f p acket ti mes th at
the CS 2000 will use, this is not recommended. By provisioning specific
packet times, the SDP packet time will no longer pass transparently
through the MCP. This means that the packet time negotiation that
would normally occur at the client level is now being handled at the
server level, and this could lead to voice connection issues.
The SDP sent by the CS 2000 media gateways is only guaranteed to
contain the v=, c= , m=, and a= SDP heade rs in the SDP message. The
gateways can accept reception of other SDP parameters, so no
screening needs to be done on the SDP that the MCP sends to the CS
2000. The only exception is the screening of the m= lines to only
include the audio codecs supported by the gateways. For the SDP that
is received from the CS 2000, the MCP fills in the missing mandatory
parameters as specified in RFC 2327. Details on the restrictions on
SDP are specifie d in the SN03 IP Gateway InterOP Requirements
document.
Recursive Search
The MCP Back-to-Back User Agent (BBUA) handles SIP 302
responses to an INVITE sent out because of a received INVITE from
the CS 2000. The BB UA sends INVITEs for each contact in the S IP 302
response. The SIP 302 response will not be passed back to the CS
2000. The INVITES fo r each contact can be sent parallel or sequ entially
depending on the mode in which the BBUA processed the initial
INVITE, which then resulted in the SIP 302 response.
Hold
SIP implement s a Hold as a re-INVITE with the connectio n information
in the SDP set to 0.0.0.0. In this release, Hold cannot be handled by all
the CS 2000 media gateways. The MCP shields the CS 2000 from
seeing Hold re-INVITE requests. The MCP, when equipped with an
RTP Media Port al, ca n ma nage the RTP connections witho ut af fe cting
the connection to the CS 2000.
Retrieve
Retrieve is also implemented in SIP through a re-invite. To retrieve a
party on hold the new invite contains va li d SDP. The new SDP is use d
to restore the media connection between the two clients.
Long Call Audit Timers
In order to prevent hung calls between the two platforms, a long call
audit timer is implemented between the two platforms. Both platforms
use the INFO ping capability described in the SIP INFO message
RFC2976. This involves sending an INFO message with no message
body. Upon receipt of this message, a client should send a 200 OK if
the call exists. If the response to the INFO is any of the following
response codes, then the platf orm sending the message assumes that
the call no longer exists and frees all resources associated with the call:
If the platform sending the INFO receives a response code other than
the ones listed above, it tre at s the re sponse a s a valid au dit resp onse,
and does not bring the call down.
The CS 2000 sends a 2 00 OK for a n em pty IN FO message and a 481
response code if the call leg does not exist.
Supported media
The following are the media su pported b y the inter working o f MCP and
the CS 2000:
•G. 711 PCMU
•G. 711 PCMA
•G. 729
•G. 723
•G3 Fax
•Modem
Note 1: This list represents a full list of the media capabilities the
MCP can use while inte rworkin g wit h the CS 2 000. The actual list of
supported media types depends on the capabilities of the gateways
that the CS 2000 is using.
Overview 9
Note 2: G3 Fax and Modem media calls are used only for CS 2000
gateway calls as there are no MCP endpoints that support either of
these two media types.
QoS
MCP implements its Quality of Service through the DiffServ
(Differenti ate d Ser vi c e) feature. The parameters of MCP QoS includ e
the following:
•QoS DiffServ code for s i gnaling: specifies signaling quality for SIP
clients.
•QoS DiffServ code for au dio: specifies audio quality for SIP clients.
•QoS DiffServ code for video: specifies video quality for SIP clients.
•QoS 802.1p for service priority: specifies service priority for SIP
clients.
Interworking with third-party gateways
The MCP can use SIP signaling between the MCP networ k and a
third-party SIP-enabled gateway. The third-party gateway provides the
necessary signalling interworking between the MCP network and the
other network to whic h the gateway is conn ected. For example, a
line-based voicemail server requires that a Line Gateway be placed
between the MCP network and the voicemail server. This Line Gateway
would process SIP message s to/from the MC P network and create the
corresponding line signalling to/from the line network. “Interworking
with third-party voicemail servers” on page 13, provides more
information about how the MCP interworks with non-SIP aware
voicemail servers.
Interworking with traditional phones
The MCP provides Con verged Desktop Servi ces (CDS) to facilit ate the
interworking with traditional phones of the TDM network. This allows
users to have a p ersonal computer (PC) provide the m ultimedia porti on
of their communication session while having the traditional telephony
system provide the voice portion of their communication session.
Functional description
A user’s Converged Desktop consists of a regular TDM telephone, and
a SIP Multimedia PC Client (PC Client) software provisioned as a
Converged PC Client. Figure 2 shows how the Converged PC Client
interconnects wit h the ne twor k. Th e C o nve rged PC C l ient pr ovide s a n
enhanced communication experience to the user, while the TDM
telephone works exactly as it does today.
CDS enhances the end user’s communication experience in a variety
of ways:
•Advanced Call Handling: The user can use the MCP Personal
Agent web pages to control the user’s availability. By providing this
ability to CDS users, features not easily accessible on existing TDM
switching systems now become viable. For example, a user can
activate MCP-based f orking (using the Personal Agent we b pag es)
so that when the user’s desktop telephone is called, the users cell
phone rings as well. Once one leg of the forked call is answered, the
other leg stops ri ng in g.
•Inbound call log: Allows the user to see who has cal l ed and when
the call occurred.
•Video calling line identification: Allows the user to see who is callin g.
The picture is retrieved from the network-based address book
accessible on the Converged PC Client.
•Redirection of incoming calls at the Converged PC Client: Upon the
arrival of an incoming call, the user may click on the “Redirect”
button, and send the incoming call to another address. Once the
user answers the call (using the TDM phone), the redirect function
is no longer available.
Overview 11
•File transfer: If both the originator and terminator support the MCP
file transfer collabora ti on ap pl ic ation , t hen files can be transferred
back and forth between the two users. The PC Client (both
Converged and non- C on ver ge d) is th e only endpoint that supports
this functionality.
•Whiteboard sharing : If both the originator and te rminator support the
MCP whiteboard collaboration application, then a whiteboard
session can be set up between the two user s. The PC Cli ent (bot h
Converged and non- C on ver ge d) is th e only endpoint that supports
this functionality.
•Clipboard transfer: If both the originator and terminator support the
MCP clipboard transfe r collaboration applicati on, then the Windows
System Clipboard may be transferred between the two users. The
clipboard transfer application allows a user to “Copy (CTRL-C)”
items such as PowerPoint slides or sections of Excel sp readsheet s
to the clipboard, and then sends them to the other party. The other
party then “Pastes (CTRL-V)” the items. The PC Client (both
Converged and non- C on ver ge d) is th e only endpoint that supports
this functionality.
•Web Co-bro wsing: If both the or iginator an d termina tor are capa ble
of this functionality, then one user can automatically drive the
other’s web browser. The PC Client (both Converged and
non-Converged) i s the only endpoi nt that support s this functionality.
The Web Client supports the reception of web pages, but cannot
send web pages to a Converged PC Client.
•Instant Messaging (IM): The Converged PC Client can send and
receive messages from any client that support s the Nortel Ne tworks
IM format. All MCP clients support sending a nd receiving of inst ant
messages with each other.
•Presence state indications: The Con verged PC Client allows the
user to select a presence state in the MCP ne twork. The Converged
PC Client also allows the user to see the presence state s for the
Buddies defined in the user’s network-based address book.
PRI is used as the interface between the existing TDM switching
system and the MCP. The MCP supports the following PRI protocol
variants:
•AT&T 4ESS(AT4)
•AT&T 5ESS (E10)
•AT&T TR 41459
•Bellcore National 2
•ETSI
•ECMA-143
•ETS 300 102-1
•Northern Telecom DMS-100 (DMS)
•NIS A211-1
•QSig
For more detai led information a bout the which PRI varian ts the SIP PRI
Gateway supports, please refer to the MCP SIP PRI Gateway Basics
document.
Successful interworking between the MCP and the TDM switch
requires that the TDM switch acti vate the Simultaneous Ring (SimRing)
feature and assign each user a SimRing number. A user in the TDM
switch that has acquired this SimRing n umber can be a C DS user . Th e
SimRing featur e must send “SimRing” calls to a routable and unique
number for each CDS user.
The MCP system operator must provision a user a s a CDS user . A CDS
user cannot use the SIP Multimedia PC Client for voice. The CDS
user’s TDM phone is u sed for voice. In addition, the CDS user m ay use
other SIP endpoints (such as the SIP Multimedia Web Client or an
i2004 controll ed by the IP Client Ma nager (IPCM) for voice over IP calls.
An MCP alias must be set up for each user so that the alia s is the same
as the Calling Line ID sent from the TDM switch to the MCP over PRI.
When a non-CDS MCP user calls a CDS user, the call is sent out the
gateway to the CDS user’s TDM phone and the non-CDS user’s
public/private charge ID is used to identify them to the TDM switch as
the calling party. This charge ID is sent to the Converged PC Client, on
the SimRing leg of the call, an d is used by the Con verged PC Cli ent to
contact the calling party’s MCP client. Therefore, the non-CDS user’s
charge ID must be included as an alias in the non-CDS user’s
provisioning. A non-CDS user’s charge ID cannot be shared amongst
users within a domain because the charge ID must be included as an
alias and a user's aliases must be unique within a domain.
Calls to a CDS user must terminate to the existing switching system of
the CDS user before the call is routed to the MCP. For example, the
originator’s existing switching system must route calls using the
existing systems, as opposed to sending the call to the MCP. This is
required since all calls from the SIP PRI gateway are implied to have
been triggered by the SimRing feature on the existing switching
system.
Overview 13
Interworking with third-party voicemail servers
There are three major types of third-party v oicemail servers. The
following sections describe how the MCP can interwork with the
following types of third-party voicemail servers:
•SIP-based voicemail servers
•Trunk-based voicemail servers
•Line-based voicemail servers
SIP-based voicemail servers
SIP-based voicemail servers are SIP-enabled and can interwork
directly with the MCP network. SIP is used to set up connections
between the client and the voicemail server. The RTP Media Portal is
used to carry the media packets between t he client and the voicemail
server.
Figure 3 shows how the MCP interconnects with a SIP-aware
third-party voicemail server.