06_IP PBX

7/21/2019 06_IP PBX

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1 © Nokia Siemens Networks CN60025EN50GLA0

For public use – IPR applies

IP PBX

FN 1957

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Challenge:

• Enable Corporate PBX connectivity in IP domain

• Modern corporate IP PBX’s are utilizing SIP and RTP for connectivity

• Signalling interfaces are evolving to SIP

Solution:

• MSS System provides interoperability with IP corporate networks

• SIP Trunking interface enables connectivity and PBX interface services for SIP PBXcustomers

• MSS System supports now both TDM and IP based PBX interfaces

• Provides same service set for SIP PBX interface and TDM PBX interface

Operator benefits

SR4.0 IP PBX support

M16.1

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Functionality:

• Different routing and charging inMSS for each individual IP PBX

– No registration from the SIP PBX tothe MSS

• Calling Line Identity (CLI)

Screening: – MSS validates the CLI sent by thePBX

– MSS expands the PBX extensionnumber to national or internationalnumber

• Calling Line Identity Restriction(CLIR) handling PBX interface:MSS removes the CLI on PBXterminating calls when CLIR is set

• Note: In Open MSS supportavailable in SR5.0 (Ma16.1)

SBC

IP BB

MGW

IP

MSS

SIP

IP PBX support

SR4.0

M16.1

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IP PBX UNI - introduction

• The IP PBX interface is perceived as SIP UNI (User Network

Interface) due to the call interconnection with private networkrequires special call procedures activated on peer Client (IP PBXwith extension ranges) bases in the MSS in order to guarantee thecalls are routable and chargeable according the call configurations.

• The IP PBX interface is a new feature in the MSS system. Theimplementation has been split in the following phases: – Phase1

▪ UNI support and CLI screening (supported by Md16.1)

– Phase2

▪ a/u-law and multi country support (supported by Md16.1)

– Phase 3 (planned)▪ MSS SR5.1 (M16.1EP2 and U6.0) => PBX registration

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Access

NetworkSBC

IP PBX S

S

PBX extension with mobile and VoIP -Optimization

PRI

M

GW

IUA30B+D

Optimized

IP PBX:

•IP PBX interconnect to

MSS/MGW via SBC(CFX5000/ ACME)

•SIP signaling

•IP PBX call control and

rooting in MSS

Traditional

ISDN PRI PBX:

•2M trunk connected to

nearest MGW (E1 TDM)

•Functionality in MSS/MGW

•‘old school’ method

MSS

MSS

MGW

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IP PBX in MSS System architecture

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Functionality

• Different routing and charging in MSS for

each individual IP PBX• Calling Line Identity Screening

• Number Normalization

• UNI CLIR Handling on terminating calls

• Ringback Tone Connection and Early

Media

• SIP “anonymous” mapping with ISUPpresentation indicator

• Call Hold and Retrieval Enhancements inNVS/MGCF

• Configurable SIP Hold and resumeindication mapping between SIP and ISUP inthe MSS

SBC

IP BB

MGW

IP

MSS

SIP

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Different routing and charging

•MGCF routing and analysis is required to enable calls to be differentiated. A

dialed number may have different routing implications, depending, forexample, on prefix numbers. Calls are therefore differentiated by theintended destination: IP PBX_1 for calls outside the IP PBX domain; IPPBX_2 for calls within a service group.

•Route level parameters are used for SIP outgoing calls when theMSS/originating-MGCF sends an INVITE request.

•Circuit group level parameters are used for SIP incoming calls when theMSS receives an initial INVITE.

•SIP CGRs are identified by the Fully Qualified Domain Name (FQDN), orfrom the IP address of the top-most Via header, if the optional trunk-group IDfeature is active and the request contains it in the R-URI or the Contact

header field.

•The MGCF provides IN trigger information for all IN Service ControlFunctions (INSCF) so that the INSCF can differentiate routing and chargingcontrol for each IP PBX.

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SIP CGR

On SIP CGR basis as an existing function the MSS either uses

a SIP port configured at the SIP message parameters or defaultport 5060.

Capacity

SIGU from MSS and ISU,NPGEP and TCU dimensioning needsto be considered.

Restrictions

The number of IEIFIL entries (thus the number of SIP CGRs) isincreased: the limit is 1500.

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CLI&RN Screening

CLI Screening is a Call Control function in which a Calling/Redirecting

identity provided by the PBX on SIP originating call is verified by the MSS.SIP caller identity can be conveyed in From, P-Asserted ID (PAI) or Remote-party-ID (RPI) headers. The identity of a redirecting party is conveyed in aSIP Diversion header.

A user identity in SIP FROM/PAI/RPI or Diversion is verified and passed or

modified to be public number which can be identified by any publicdestination networks.

For an examplePBX_1 number area = 9622xxxPBX_1 direction number/call number = 96221The extension number ranges = 100-300 and 600-800

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PBX_1 originatingcall to CS CN

CLI screeningaccepts theextension (callnumber body +extension 300)

PBX_1 / SIP

INVITE: … From:"9622300"<sip: [email protected]>Remote-Party-ID:

<sip:[email protected]>;privacy=off…..

PSTN ISUP/BICC

IAM:… CALLING PARTY NUMBER: Digits: 9622300nature of address ind: nat

presentation ind: allowedscreening ind: user-provided, verified and passed…

PBX_1 originatingcall to CS CN

CLI screening doesnot accept theextension (callnumber body +extension 500).The call number96221 is set as CLI.

PBX_1 / SIP

INVITE … From: "9622500"<sip: [email protected]>Remote-Party-ID: <sip: [email protected]>;

privacy=off…..

PSTN ISUP/BICC

IAM:

… CALLING PARTY NUMBER: Digits: 96221nature of address ind: natpresentation ind: allowedscreening ind: network verified…

PBX_1 forwardingcall to CS CN

CLI screening doesnot accept theextension (callnumber body +extension 500).

The call number96221 is set as RD.

PBX_1 / SIP

INVITE … From: “0504069072"<sip:[email protected]>Remote-Party-ID: <sip:[email protected]>;privacy=off

Diversion: <sip:[email protected]>; privacy=none; counter=1;reason=unconditional…..

PSTN ISUP/BICC

IAM: ….CALLING PARTY NUMBER:Digits: 504069072, nature of address ind: nat,presentation ind: allowedscreening ind: user-provided, verified and passed

REDIRECTING NUMBER:Digits: 96221, nature of address ind: national,presentation ind: allowed

ORIGINAL CALLED PARTY NUMBER:Digits: 9622500, nature of address ind: nat,presentation ind: allowed

REDIRECTING INFORMATION: redirecting reason=unconditionalcounter=2….

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Number Normalization

•With NN, in PBX originated SIP calls, the Calling party identities and thecalled number can be modified in to national, international or local formataccording the MGCF configuration

•The received called and calling numbers can be national, international, orthe format of the number can be not known.

•Numbers such as Called Party Number (SIP Request URI), Calling PartyNumber (SIP From/RPI/PAI), Redirecting Number (SIP top most Diversion)and Original Called Number (SIP bottom most Diversion) may benormalized.

•Number normalization and CLI screening can be clearly interrelated. A

number is first screened then modified by number normalization to theappropriate format

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UNI CLIR

• At an IP PBX terminating call when CLIR is required

(received from incoming side) concerning CLI, RD and the MSSdoes not send the number information to the IP PBX.

•The behavior of the MSS ensures the number is not presentedfor the IP PBX user.

•In the case of that the MSS sends the URI of the header(From, RPI, PAI, Diversion) as “Anonymous” to the IP PBX.

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CS CN forwarding callto PBX_1

UNI CLIR is activatedwith full hiding

PSTN ISUP/BICC

IAM: ….CALLING PARTY NUMBER:Digits: 405069072, nature of address ind: nat,presentation ind: restricted

REDIRECTING NUMBER:Digits: 405069073, nature of address ind: national,presentation ind: restricted

ORIGINAL CALLED PARTY NUMBER:Digits: 405069073, nature of address ind: nat,presentation ind: restricted

REDIRECTING INFORMATION: redirecting reason=unconditionalredirecting counter=1….

PBX_1 / SIP

INVITE:….From: "Anonymous" <sip: [email protected]>

Remote-Party-ID: <sip: [email protected]>; privacy=full;

Diversion:

<sip: [email protected]>; privacy=full; counter=1;reason=unconditional…

CS CN forwarding callto PBX_2

UNI CLIR is activatedso that receivedidentities in questionare sent with privacy

PSTN ISUP/BICC

IAM:

….CALLING PARTY NUMBER:Digits: 405069072, nature of address ind: nat,presentation ind: restricted

REDIRECTING NUMBER:Digits: 405069073, nature of address ind: national,presentation ind: restricted

ORIGINAL CALLED PARTY NUMBER:Digits: 405069073, nature of address ind: nat,presentation ind: restricted


PBX_2 / SIP

INVITE:….From: "Anonymous" <sip: [email protected]>

Remote-Party-ID: <sip:[email protected]>; privacy=full

Diversion:

<sip:[email protected]>; privacy=full; counter=1;reason=unconditional…

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SIP Anonymous mapping with ISUP presentationindicator

•IP PBXs widely use an assumed name in the display-name component of

the From and Diversion header field as explained in the IETF RCF3323 forthe privacy.

•It is RECOMMENDED that user agents seeking anonymity use a display -name of "Anonymous". The MSS supports mapping between a display -name of "Anonymous" or “Anonymous” URI and the ISUP presentationindicator for the interconnection calls.

•The “Anonymous” information is mapped:

from the SIP From field to the ISUP Calling Party Number/ACgN,

from SIP top most Diversion to the ISUP Redirecting Number, and

from the SIP bottom most Diversion to the ISUP Original Called Number

E l

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PBX_1 forwardingcall to CS CN

A forwarding callwith Display“Anonymous”.

Caller Identity is onlyreceived at SIP From

PBX _1/ SIP

INVITE:…

From:“Anonymous"<sip: [email protected]>

Diversion:"anonymous"<sip:[email protected]>;reason=unconditional;….

PSTN / ISUP/BICC

IAM:

… CALLING PARTY NUMBER:Digits: 9622300nature of address ind: nat

presentation ind: restricted screening ind: user-provided, verified and passed

REDIRECTING NUMBER: digits: 9622301 (Note that the CLI screening accepts thenumber)nature of address ind: natpresentation ind: restricted

REDIRECTING INFORMATION:redirecting reason=unconditionalcounter=1…

CS CN forwardingcall to PBX_1

A forwarding callfrom PSTN.UNI CLIR is activatedwith full hiding

Caller Identity is onlysent at SIP From

PSTN ISUP/BICC

IAM: ….CALLING PARTY NUMBER:Digits: 405069072, nature of address ind: nat,presentation ind: restricted

REDIRECTING NUMBER:Digits: 405069073, nature of address ind: national,

presentation ind: restrictedORIGINAL CALLED PARTY NUMBER:Digits: 405069073, nature of address ind: nat,presentation ind: restricted


PBX_1 / SIP

INVITE:….From:"Anonymous" <sip: [email protected]>

Diversion:

<sip: [email protected]>; privacy=full;

counter=1; reason=unconditional…

Example

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Ring back tone connection and early media

• An IP PBXs might connect some early media on a setup

phase of a call. The early media here refers to media e.g. callwaiting information, announcement or tone, which isgenerated by an IP PBX at ringing phase of the call.

• On an IP PBX terminating call an early media from an IP PBX

can not be connected through at ringing phase due to theMSS/MGCF and MGW always connects ring back tone to aCaller.

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SIP Call Hold and Retrieval enhancements

•Call Hold is handled differently in the Telco world to the IT

world. In the IP PBX enterprise solution, the MGCF has theability to support both call hold models.

•The MSS Call Hold handling logic is enhanced so that it canidentify the different cases/models and behave according to the

appropriate standards.• Concerning the IP PBX enterprise most of the SIP devices

behave according IETF RFC3264 (SIP Offer/Answer model)

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For an example: Upon call hold an IP PBX responses with mediainactive

MSS/MGCF IP PBX

Call hold is initiateda=sendonly

IP PBX responds with

a=inactive but doesn't want

to indicate Double Call

Hold

a=inactive

MGCF sees it as Double Call Holdbut IP PBX doesn’t want to indicate

Call Hold.

a=sendrecv

SBC is stateless, so it just

forwards SDP.

Current: a=recvonly

Solution: a=sendrecv

New requirement:

MGCF has to be prepared for a =

inactive response.

Call Retrieval

Currently, the call is released in this phase.

Not allowed response by RFC3264.Normally, a=sendrecv

shall make the Call

Retrieval

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Hold/Resume mapping between SIP andISUP/BICC as configurable option at the MSS

The problem here related to an attendant service at IP PBX in

which there might be so many hold/resume requests at a callthat the hold/resume indication at a mobile phone (or PSTNphone) begins to be disturbing for the mobile user at the call.Therefore it was requested the MSS shall support as an optionthe hold/resume mapping between ISUP and SIP.

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For an example: An IP PBX provides hold and resume

IP PBX MSC-S/MGCF MGW CS CN

INVITE/UPDATE (SDP Offer a= sendonly or c=0.0.0.0)

CPG (Remote Hold)

H.248: MOD.req [Context ID = C1, Termination ID = T1, ALL, Isolate]

H.248: MOD.resp [Context ID = C1, Termination ID = T1]

200OK (SDP Answer)

Change Through-

Connection=inactive

ISUP CPG is either sent or not sent according

the control of new SIP CGR parameter.

By default, the CPG is sent. ACK (If INVITE was used)

INVITE/UPDATE (SDP Offer a= sendrecv or c=IN IP4 224.2.0.1/127)

H.248: MOD.req [Context ID = C1, Termination ID = T1,T2, bothway]

H.248: MOD.resp [Context ID = C1, Termination ID = T1,T2]

CPG (Remote retrieval)

Change Through-

Connection=both

ISUP CPG is either sent or not sent according

the control of new SIP CGR parameter.

By default, the CPG is sent.

200OK (SDP Answer)

ACK (If INVITE was used)

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Multi country support for IP PBX interoperability

The multi country support means that the MSS supports two or more SIP

enterprise networks (IP PBXs) which are located in different countries.This affects Routing and Charging analysis, CLI Screening and numbernormalization which are needed to manage here on country specific.

▪The functionality introduces the possibility to set a country code, anational prefix, an international prefix and a trunk code in the CallControl parameter set.

▪ This helps identify the origin of the received A number (CLI,RDN,OCN) incase of an incoming call (CGR) and also the formatting of the sent number incase of an outgoing call (Route).

▪The feature is only available after the activation of the feature license. Thelicense feature code:1972.▪For example

Incoming side: -> The Incoming Call Control parameter set is attached in CGR

ZCPM:NCCP=EXTIP:::::::INTPRFX=003,NATPRFX=03,CCODE=122,TRUNKCOD=49,NUMNORM=Y;

Outgoing side -> The outgoing Call Control parameter set is attached in Route

ZCPF:NCCP=BASICOUTPSTNPBX::CCODE=122,NUMNORM=Y,TRUNKCOD=49;

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HW and SW requirements

Hardware

The feature functions in DX MSS.

While there are no specific hardware requirements it should benoted that SIGUs/CCSUs are used not PAUs.

A properly configured DNS is also recommended.

Software

FN1972 SIP IP PBX needs to be activated

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Statistics

•In Traffic Category Measurement, two new category counters

for the IP PBX originating and terminating calls are introduced.•Inclusion of the counters in the reports is optional.

•This optionality is controlled by the SIP_IP_PBX_SUPP FIFILEparameter.

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Charging

A number of changes to CDR fields are introduced.

DRIECTORY_NUMBER

•Every IP PBX has a CLI screening set assigned. The "Directory Number"field in the Charging record holds the AreaCode+DirectoryNumber (oroptionally the Country-Code+AreaCode+DirectoryNumber) of the IP PBXthat is defined in the screening set.

•This way the IP PBX to be charged can be identified. This field is availablein the PBXO,PBXT, SOC and STC CDRs.

SIP_SIG_MODE

This field contains the mode of the SIP call. The field is available in the SOCand STC CDRs. The field has the following additional value:

• 07H SIP PBX

A maximum of 1500 SIP CDRs can be configured in an MSS for IP PBXs.

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Call bases G.711 a - law or G.711 u-law inMGW for SIP

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Challenges

• When MSS is deployed as international GW, interworking

between different international standards for G.711 codec (a-law/u-law) may be required.

• Some network elements (e.g. IP-PBX) support only either a-

law or u-law (both variants are delivered to ANSI and ETSImarkets)

• Furthermore there are also some VoIP devices which require

modifying the G.711 law during the call e.g. when anintermediate announcement is given .

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IP-PBX Codec conversion in MGW

MGWIP

MSS

SIP

a-law

u-law

SIP

SIP

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Solution to meet challenges

• MGW (MSC Server system) supports the simultaneous usage

of G.711 a-law/u-law codec in the same network elements.

• The MGW supports transcoding between G.711 a-law and u-law.

Main use cases:

• international interconnect

• using a “foreign” Network element in the home network (e.g. all the elements and interconnected

elements are using a-law, but there is one box in the network using u-law (e.g. the IP PBX case)

Operator benefits

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Functionality

• The MSS controls the MGW (MGCF) to reserve DSP

resources for the selected codec on a call-by-call basis byoverriding the existing G.711 law.

• a/u-law selection is done on User Plane Destination basis.

• It is possible to select either G.711 a-law or u-law accordingthe SDP negotiation (SIP Offer/Answer).

• It is possible to re-negotiate G.711 a-law or u-law (SIPOffer/Answer) during the call.

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Restrictions

•Conversion is supported only in IP-Backbone (Nb) and IP

Access (Mb) interfaces when call control protocol is SIP-I orSIP.

• Not supported interfaces: A, Iu, PSTN, Nb (BICC)

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Dependencies and sales items

Release dependencies:

This Sales Item is available as an extension on top of the MSS/IP PBX UNI

(N.4739) or on top of the other SIP/SIP-I/NVS Sales Item.Licenses for MGW:

– 3GNLIC0076 MGW A-u law conversion for SIP

▪ pricing trigger: Number of IP Access ports

–3GNLIC0078 MGW A-u law converse for SIP internet

▪ pricing trigger: All connections

Licenses for MSS:

– MSGINV61504 C.1504 G.711 a/u-law in MSS LTU

▪ Pricing trigger: VLR subscribers during BH

– MSGINV21508 C.1508 G.711 a/u-law in MSS (IPPBX) LTU

▪ Pricing trigger: simultaneous call

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Statistics

•The selected codecs are also shown in the MSC Observation report, Codec

measurement, and CPI MML printout***•Transcoding measurement in Open MGW (912) includes new counters forthis feature

• A_U_LAW_CONVERSION counter: The number of usage of G.711 codec(a-law or μ-law) and configured encoding law in MGW is different than MSS

requests.• A_U_LAW_CONVERSION_ERROR counter: The number of unsuccessfulG.711 conversion from a-law to μ-law, or vice versa

*** Needs to be rechecked

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Charging

•The codec information is sent to charging for generating

Charging Data Record (CDR).•If the codec is changed during the actual call, chargingmodification (intermediate charging) is also triggered. Theusage of new voice codecs are also recorded in Charging Data

Records (CDRs).

C C G G

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FC006460 Call bases G.711 a-law or G.711 µ-law inMGW Functionality and operator interface 1/2

The MGCF and MGW support on call basis SDP negotiation and

modification concerning encoding law of the G.711 codec. Furthermore theMGW supports “transcoding” between the G.711 a-law and G.711 µ-law.

It is important to take account the restriction that conversion is supportedonly in IP Backbone (Nb) and IP Access (Mb) interfaces when call control

protocol is SIP-I or SIP.

The MGW level parameter defines the default encoding law for MGW.

•The parameter is mgw common pcm-standard.

•This is the only option in A, Ater, TDM and Nb (BICC) interfaces.

TFO functionality can be used simultaneously with“transcoding” between the G.711 a-law and G.711 µ-law.

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