HD VOICE

.
High-Definition Voice – What
is all about ?

› Raising the quality of voice communication in mobile

networks with more natural sound and improved
intelligibility in order to:
- Motivate end users for longer and more frequent voice calls
- Give competitive edge to operators based on improved
speech quality
- Attract enterprise users through improved quality of service
(teleconferences, speech recognition)
- Enable the same user experience in Mobile and Fixed
Networks

HD Voice | 2012-09-24 | Page 2 (59)

Why HD voice in 13A?
› Strategic fit
– Support of HD Voice for IP Interconnect will increase AMR-WB sales both
in RAN and CN since it increases the number of HD voice calls in total
– HD voice is a strong driver for introduction of IP Interconnect generating
also sales of SIP/SIP-I and VoIP features

› Customers
– Orange/FT, Bouygues, Rogers, TIM, Swisscom, Telstra, TurkCell, …
– Orange France FOA is in preparation

› Competitors
– NSN and Huawei support AMR-WB towards SIP and SIP-I

HD Voice | 2012-09-24 | Page 3 (59)

HD voice use cases
HD voice between mobile
operators

› SIP-I is used for interconnection between two mobile networks

› The MSC-Ss use end-to-end codec negotiation to select AMR-WB
› The M-MGws are in the TrFO mode

HD Voice | 2012-09-24 | Page 5 (59)

IMS interwork - HD Voice
towards IMS/volte

› SIP is used for interconnection towards IMS.

› The MSC-S uses end-to-end codec negotiation to select AMR-WB
› The M-MGw is in the TrFO mode
HD Voice | 2012-09-24 | Page 6 (59)
Ims Break-out via IP POI

› MSS is used as IP POI towards mobile networks using SIP-I

› The MSC-S uses end-to-end codec negotiation to select AMR-WB
› The M-MGw is in the TrFO mode

HD Voice | 2012-09-24 | Page 7 (59)

HD voice FEATURE SET AND
REQUIREMENTS

› HD Voice feature set in MSC 13A is introduced through the

following requirement specifications:
– End-to-End Codec Negotiation for Interworking with SIP/SIP-I
– HD Voice with AMR-WB Speech Codec
– SIP Preconditions

HD Voice | 2012-09-24 | Page 8 (59)

HD VOICE RELATED
standardization

› 3GPP (R10) TSs:

– 29.163, Interworking between the IP Multimedia (IM) Core Network
(CN) subsystem and Circuit Switched (CS) networks
– 24.229, IP multimedia call control protocol based on Session
Initiation Protocol (SIP) and Session Description Protocol (SDP);
Stage 3
› IETF RFCs:
– RFC 4867, RTP Payload Format and File Storage Format for the
Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
(AMR-WB) Audio Codecs
– RFC 3312, Integration of the resource Management and Session
Initiation Protocol (SIP)

HD Voice | 2012-09-24 | Page 9 (59)

END-TO-END CODEC
NEGOTIATION FOR
INTERWORKING WITH SIP/SIP-I
SCOPE
› End-to-end codec negotiation (OoBTC procedures) and HD
voice was already possible in CS core network
› Capability is now introduced for end-to-end codec
negotiation with SIP(-I) based networks and HD voice
across network borders is enabled
WCDMA / GSM
RAN

HD voice
HD voice
HD voice
IP
CS Core SIP/SIP-I Interconnect
Network or
WCDMA / GSM IMS network
RAN

MSC-S

HD voice

HD Voice | 2012-09-24 | Page 11 (59)

traffic cases

› Calls from CS Core Network to IMS network or networks

using IP interconnect (PSTN, PLMN, IP-PBX, IVR, VMS):
– 2G/3G originating access to SIP or SIP-I
– BICC to SIP or SIP-I
› Calls from IMS network or networks using IP interconnect
(PSTN, PLMN, IP-PBX, IVR, VMS) to CS Core Network:
– SIP or SIP-I to 2G/3G terminating access
– SIP or SIP-I to BICC
› SIP(-I) transit call cases (IP-POI – IMS)

HD Voice | 2012-09-24 | Page 12 (59)

capabilities

› All 3GPP codecs currently supported by SIP(-I)* and

G.729** can be negotiated end-to-end
– * GSM EFR, AMR(set1), AMR-HR(set1), AMR(set7), PCM
– ** only in SIP(-I) to SIP(-I) call cases
› Flexibility to force pre-13A codec negotiation behavior per
outgoing SIP(-I) trunk
› Flexibility to further restrict node level defined SIP(-I) codec
list on a per SIP(-I) trunk basis

HD Voice | 2012-09-24 | Page 13 (59)

Feature view
New feature:
Transcoder Free
TrFO Interworking
Operation (TrFO)
MSC-S with SIP/SIP-I

MGCF Interworking Compressed Speech

with IMS on Nb Interface
M-MGW

or

MGCF Interworking
with SIP-I network VoIP Gateway

X Y

Feature X requires support of feature Y

HD Voice | 2012-09-24 | Page 14 (59)

HD voice with amr-wb
speech codec
scope
› Support AMR-WB codec for interwork with SIP or SIP-I
based networks (IMS interwork or IP interconnect
scenarios)

WCDMA / GSM
RAN

HD voice
HD voice
HD voice
IP
CS Core SIP/SIP-I Interconnect
Network or
WCDMA / GSM IMS network
HD voice
RAN

MSC-S

HD voice

HD Voice | 2012-09-24 | Page 16 (59)

capabilities

› AMR-WB (Set 0) configuration with rates 12.65, 8.85 and

6.60 kbps
› Flexibility to restrict AMR-WB usage on a per SIP(-I) trunk
basis
› Route level counter for the number of SIP/SIP-I calls with
AMR-WB payload type applied during call setup.

HD Voice | 2012-09-24 | Page 17 (59)

Feature view

Enhanced feature:
AMR-WB Speech
AMR-WB Speech
MSC-S M-MGW

TrFO Interworking
with SIP/SIP-I

X Y

Feature X requires support of feature Y

HD Voice | 2012-09-24 | Page 18 (59)

e2e codec negotiation (BICC–
SIP/SIP-I)

BICC SIP/SIP-I
MSC-S/MGCF

IAM (SCL1) MGW

Reserve IMS Connection Point

(send preferred codec (1st from the
SDP payload list) to MGW in Local
Descr.) INVITE (SDP Offer {SCL2})

18x (SDP Answer {SC2})

Configure IMS Resources (send selected
codec in Remote Descr. and, if different than
one previously sent, in Local Descr. again) MGW

APM (SC1, ACL)

COT

ACM
200 OK (INVITE) (SDP Answer {SC2})
ANM
ACK

BICC to SIP or SIP-I interworking and single payload type in SDP answer in non-reliable provisional response

HD Voice | 2012-09-24 | Page 19 (59)

 OPTIMAL MGW SELECTION
MSC/MGCF

i/c BICC TRAM o/g SIP

Existing OIP parameter “Media-

Gateway Group” (MGG2) is be IAM
OIP IAM
added to the OIP IAM message (incl. MGGr, MGG2, OIP IAM
COT on previous) (incl. MGGr, MGG2,
to carry incoming MGG. COT on previous)

MGW selection & Incoming MGG, together with

termination seizure
outgoing MGG is used to select
INVITE MGW for outgoing side.
(SDP offer)

180 Ringing
OIP BIM (SDP answer with SC)
OIP BIM (MGGp, Any MGW,
(MGGp, Any MGW, O-BCU-ID, SC)
O-BCU-ID, SC)
MGW selected on outgoing side MGW selection &
is selected also on incoming termination seizure

side (if possible).

OIP BIM
APM (I-BCU-ID) OIP BIM
(I-BCU-ID)

COT
OIP COT
OIP COT

OIP ACM
OIP ACM
ACM

Figure 1 Optimized MGW selection signaling sequence

HD Voice | 2012-09-24 | Page 20 (59)

 e2e codec negotiation (SIP/SIP-I–
SIP/SIP-I)
SIP/SIP-I SIP/SIP-I
MSC-S/MGCF

INVITE (SDP Offer {SCL1}) MGW

Reserve IMS Connection Point

(send preferred codec (1st from the
SDP payload list) to MGW in Local
Descr.)
INVITE (SDP Offer {SCL2})
Configure IMS Resources (send HP
codec* from SDP answer in Remote 18x (SDP Answer {SCL3})
Descr. and, if different than one
previously sent, in Local Descr. again)
MGW

Reserve IMS MGW

Connection Point and
Configure Remote
Resources (send
selected codec to MGW PRACK (SDP Offer {SC2})
both in Local and
Remote Descr.) 200 OK (PRACK) (SDP Answer {SC2})

183 Session Progress

(SDP Answer {SC1}) 200 OK (INVITE)
18x
ACK
200 OK (INVITE) (SDP Answer {SC1})

ACK HP codec – highest priority payload type from

SDP answer that was present in SDP offer

SIP or SIP-I to SIP or SIP-I interworking and only outgoing side supports reliable provisional response (multiple payload types in SDP answer)

HD Voice | 2012-09-24 | Page 21 (59)

INTRODUCTION of SDP Codec
List and SDP Single Codec
SIP/SIP-I TRAM SIP/SIP-I

Filtering of non activated codecs

1. INVITE
(SDP offer)
Creation of SDPCL from codecs received in Filtering of non activated
SDP offer codecs from SDPCL,
Payload type translation adding of activated codecs
to SCL not received in SDPCL
2. OIP-IAM 3. OIP-IAM
(SCL, SDPCL) (SCL, SDPCL)
4. INVITE
(SDP offer)
Translation of SDPSC to Translation of selected codec to SC and SDP
selected codec SC 5. 183 Session Progress
(SDP answer)
6. OIP-BIM 7. PRACK
9. OIP-BIM (SC, SDPSC)
(SC, SDPSC) 8. 200OK (PRACK)
10. 183 Session Progress
(SDP answer)
11. OIP-BIM
13. PRACK 12. OIP-BIM

14. 200OK (PRACK)

New OIP parameters SDP Codec List and SDP Selected Codec are introduced. The motivation was to
improve translation of payload types between SDP offers / answers for SIP/SIP-I to SIP/SIP-I
interworking cases. This also enables TrFO when there is no equivalent representation in existing
OoBTC codec lists (e.g. G.729).

HD Voice | 2012-09-24 | Page 22 (59)

SIP Preconditions
scope

› Support for segmented SIP QoS (Quality of Service)

precondition function allowing SIP UAs to suspend session
establishment until a set of QoS preconditions are met
› Enhanced MSC-S interworking capabilities towards SIP or
SIP-I networks, especially in case when end-to-end codec
negotiation is used

HD Voice | 2012-09-24 | Page 24 (59)

Intro

› Introduction of end-to-end codec negotiation enables

reaching the B-party without reservation of the bearer
resources
› Consequently, ‘media clipping’ and ‘ghost ringing’ are
possible

Preceding BICC SIP/SIP-I Succeeding

Network MSC-S Network

IAM (SCL1)
INVITE (SDP Offer {CL1})

183 (SDP Answer {C1})

Bearer is not established
180 Ringing
APM (SC1)
B-subscriber alerted, inband included
COT

ACM
HD Voice | 2012-09-24 | Page 25 (59)
goalS

› Ensure bearer resources are reserved before B-party is

reached
› Improve call quality by preventing ‘media clipping’ and
‘ghost ringing’

Preceding BICC SIP/SIP-I Succeeding

Network Network
MSC-S

IAM (SCL1)
INVITE (SDP Offer {CL1},
precons not met)
183 (SDP Answer {SC1},
APM (SC1, ACL) precons not met)

COT
UPDATE (precons met)

180 Ringing

ACM
B-subscriber alerted, inband included

HD Voice | 2012-09-24 | Page 26 (59)

capabilities

› Activation on node level

› Flexibility to restrict preconditions usage on a per outgoing SIP(-I) trunk
basis
› Flexibility to define direction attribute (inactive or sendrecv) in initial
INVITE when signalling ‘preconditions not met’
› Precondition met timer’ introduced – waiting for ‘preconditions met’
indication from preceding peer (timer value is configurable on node
level)
› Route level counter for number of initial INVITE requests rejected due
to precondition failure.
› Event reported when sending or receiving 421 Extension Required final
response (preconditions used but support for preconditions or reliable
provisional responses not indicated).

HD Voice | 2012-09-24 | Page 27 (59)

Feature view

Enhanced feature:
MGCF Interworking
MSC-S with IMS

Enhanced feature:
MGCF Interworking
with SIP-I network

HD Voice | 2012-09-24 | Page 28 (59)

SIP PRECONDITIONS USED
BETWEEN TWO MSC-S
A B
a=inactive INVITE local
a=curr:qos local none
a=curr:qos remote none
(Supported: precondition, 100Rel)
remote
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv a=inactive
183 a=curr:qos local sendrecv
a=curr:qos remote none
(Required: precondition, 100Rel) a=des:qos optional local sendrecv
a=des:qos mandatory remote sendrecv
a=conf:qos remote sendrecv

Resource Reservation PRACK

200 OK PRACK
local
remote Preconditions met at the caller

a=sendrecv
a=curr:qos local sendrecv UPDATE
a=curr:qos remote none local
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv remote
a=sendrecv
200 OK UPDATE a=curr:qos local sendrecv
a=curr:qos remote sendrecv
a=des:qos optional local sendrecv
a=des:qos mandatory remote sendrecv

180 (Ringing)

Resource reservation needed only at A side. Resources reserved

(precondition met) at A side after sending PRACK
HD Voice | 2012-09-24 | Page 29 (59)
EXCLUDED SCOPE

› The following items did not fit 13A timeframe and are planned to be
implemented in 14A:
- Routing based on wideband codec. It should be possible to select
outgoing route with IP transport based on the wideband (AMR-WB or
G.722) codec as highest prioritized codec received on incoming side.
Interesting for operators with mixture of IP and TDM transport in the
network.
- HD Voice with support of G.722 codec. This would be new
commercial optional feature. Enhanced AMR-WB Speech and TrFO
interworking with SIP and SIP-I are prerequisite, of course.

HD Voice | 2012-09-24 | Page 30 (59)

DESIGN DETAILS
Overview of impacted
system modules
(G)MSC /TSC-S (BC)
OIP OIP
DBQAM MAGAM
OIP OIP
SSFAM ISUP
ISUP
OIP
TRAM OIP
SIPTA
SIP/SIP-I
-I TUP
OIP TUP
-I
INTRO
INTROIP OIP
OIP
IUSAM
DSS1
BICC
BICC

BASE BASE BASE RTDMASE

RMP
GACCSE GACCSE
BASRM CONRM RTARM

Impacted internal protocol or interface Not impacted external protocol

Impacted external protocol Impacted SM

HD Voice | 2012-09-24 | Page 32 (59)

Impacted system
modules/blocks in MSC-S
XSS/TSS
BICC: BIPHT
SIPTA: SIPHIT, SIPHOT, SIPBC, SIPCO

IUSAM: ADJI, ADJO

CONRM: GADESCR, GALINK
TRAM: INTRO
MAGAM: MTA, MUAMCO, MRRASG
+APZ VM: ACSIP (Clayton block)

HD Voice | 2012-09-24 | Page 33 (59)

Main Development
Challenges
› New MGW selection handling for e2e – outgoing SIP/SIP-I termination is
now seized before incoming one, but MGW selection must remain optimized.

› Additional queuing handling for outgoing SIP or SIP-I – since now initial
INVITE is sent before actual through-connection – all alerting/answering
messages must be queued until through-connection is achieved.

› New SIP/SIP-I signaling sequence due to introduction of e2e codec

negotiation and SIP Preconditions

› Support of new AMR-WB payload type on SIP/SIP-I

› Introduction of transparent transfer of SDP Payload Type list and

Selected Payload type through MSC-S – to optimize e2e codec negotiation
for SIP/SIP-I to SIP/SIP-I traffic cases

HD Voice | 2012-09-24 | Page 34 (59)

VERIFICATION DETAILS
Main Verification Traffic
Cases
› 2G or 3G Mobile Originating/Terminating to SIP,
› 2G or 3G Mobile Originating/Terminating to SIP-I,
› PRA to SIP,
› PRA to SIP-I,
› ISUP to SIP,
› ISUP to SIP-I,
› BICC to SIP,
› BICC to SIP-I,
› SIP to SIP,
› SIP to SIP-I,
› SIP-I to SIP-I
› In addition: interaction with Rerouting before or after result of codec
negotiation, ISDN call forwarding, CAMEL/IN services interaction, Call
Hold
› Feature interactions: e2e codec negotiation + SIP Preconditions, CMN,
SRVCC Handover, MMTel, T.38 Fax.

HD Voice | 2012-09-24 | Page 36 (59)

FUNCTION TEST - CONFIGURATION
› Function test is performed in Simulated Environment (STE).
› Test configuration includes two MSC Server Dual Blades (MSC-S DB). It is
based on standard ETSI T1 Topology. All other surrounding nodes is
simulated with STN or TTCN Additional DT needs to be loaded for SIP for
SEA-TTCN communication
STE
BSS / RNS

RANAP/
BSSAP Tool: GSM/WCDMA
simulation
MSC-S DB / GCP (IP) MGW
HLR (STN)
(SEA)
Tool: SIP/SIP-I simulation
SIP / SIP-I
SIP / SIP-I
ISUP / BICC Tool: ISUP/BICC simulation

MSC-S DB ISUP / BICC

GCP (IP) MGW
(SEA) (STN)

HD Voice | 2012-09-24 | Page 37 (59)

FUNCTION TEST - TOOLS
› SEA
› MGW simulated with (STN)MGW
STN tools currently cover:
› MGW simulation
› BSS/RNS + MS
GSM/WCDMA simulated with STN tools
› SIP/SIP-I/BICC simulated with TTCNv3
› Wireshark

HD Voice | 2012-09-24 | Page 38 (59)

FUNCTION TEST – CONTINUOUS
INTEGRATION

› HD Voice was also pilot feature for Auto LT on GEOFF

› There was a great focus on test case automatization during
test preparation – almost 900 automated test cases on
simulated environment is prepared
› Couple of hours are needed to execute regression test
› Among these, around 100 test cases is eventually added to
Auto LT Suite on GEOFF

HD Voice | 2012-09-24 | Page 39 (59)

Feature Integration and
Verification - SCOPE
› Feature testing including integration test under load and
feature disturbance testing.
› FUP Verification for intermediate and final LSV deliveries
(LFD was 1232). Main upgrade path 14.1 -> 13A is only
covered.
› Other verification related to NFR requirements, including
Characteristics Measurements for most critical traffic
cases. Results and analysis are given in feature
Characteristics Form
› Joint tests with OSS-RC for the new statistic counters
› Preparation or ERP, DCG, TSG, FCD, Demo Description
and External Test Report according to PIDS / BUGS
requirements

HD Voice | 2012-09-24 | Page 40 (59)

Feature Integration and
Verification - CONFIGURATION
› Hybrid test configuration is used including both MSC-S DB and MSC-S BC (NW3 in ETK,
it is based on standard ETSI T4 Topology) in following configurations:
1. MSC-S, APZ 212 60/1, APG43
2. MSC-S BC, APZ 214 03/1, APG43
Additional DT for VoIP and PRA needs to be loaded on top of standard DT.

STP
BSS / RNS

RANAP/ Tool: GSM/WCDMA

MSC-S DB / BSSAP
simulation
HLR GCP (IP) MGW
(Sim)
SIP / SIP-I Tool: SIP/SIP-I simulation
DNS SIP / SIP-I
SIPMux ISUP / BICC

Tool: ISUP/BICC simulation

ISUP / BICC
MSC-S BC GCP (IP) MGW
(Sim)

HD Voice | 2012-09-24 | Page 41 (59)

Feature Integration and
Verification - TOOLS
› MGW simulated with MgwSim

› GSM/WCDMA simulated with TitanSim/SIMU

› SIP/SIP-I signaling is simulated with:

- DCT 2000 (used for legacy traffic)
- SIPp (new test cases)
- OpenSIPS

› SIPmux: For SIP/SIP-I routes SIP multiplexer is used in order to enable

multiple SIP routes between 2 MSCs

› HMG: Simulation of PRA access

› DNS (Bind9)

› Wireshark

› RQM: Feature Integration test cases are stored in RQM tool, see link
https://rqm.ericsson.se/jazz/web/console/DUCI_MSS_HD_Voice
HD Voice | 2012-09-24 | Page 42 (59)
NETWORK INTEGRATION TEST
(NIT)

› Scope of NIT was feature testing for realistic e2e call

scenarios, including real nodes. In addition, preliminary
Quality of Speech measurements are performed with help
of RVC team.

› NIT is performed in RVC network in LMF normally used for

Quality of Speech measurement. (This NW is not based on
standard topologies). One additional MSC-S BC node in
ETK was also remotely connected to that network. LTE
equipment was not available at a time.

HD Voice | 2012-09-24 | Page 43 (59)

NIT-NW CONFIGURATION

VA
MSS HLR
Analysis& LTE HS
IMS
Attack
S
Tools HS
S

DN
CSCF S
MSC-BC MSC-S
BC2 MSC35

MME
SMSC
Gs
SGs MGC MTAS

SBG
SGC-XX
BGF-242 S3
Gb MRFP
IuPS SGSN CGW IM-MGW
MGW-242
MGW-242
M-MGW M-MGW
MGW-242 MGW-35
A15
S1
VA
GGSN A15 Analysis& Attack
S1-U
SBG Tools
SGC-XX
BGF-242

BSC BSC RNC RNC

BSC31 BSC39 RNC35 RNC36

BTS BTS nodeB nodeB eNodeB

TE4 Handover box

IP network

HD Voice | 2012-09-24 | Page 44 (59)

NIT-TOOLS

› Kamailio simulating SIP server

› LTE Dongle + PC with Movial SIP client
› PC with Movial SIP client
› DTP, test phones.
› CCN-R3 Handover Box
› NetEM for IP Network disturbance
› Wireshark
› SQiSE used to generate reference sound files

HD Voice | 2012-09-24 | Page 45 (59)

KEY dOCUMENTATION
FCS PHASE

Requirement Specifications:
› HD voice with AMR-WB speech codec, MSC 13A
› End-to-end Codec Negotiation for Interworking with SIP/SIP-I, MSC-S
R15.0
› Handling of SIP Preconditions, MSC 13A
Modeling Frameworks:
› MF: HD voice with AMR-WB speech codec, MSC13A
› MF: End-to-end CODEC negotiation for Interwork with SIP/SIP-I
Test Framework:
› HD Voice with AMR-WB
Note: TF covers all Sub-features. There is no MF for SIP Preconditions!

HD Voice | 2012-09-24 | Page 47 (59)

FIP PHASE (DESIGN)
Feature Design Specifications (FDS):
› FDS: TrFO Codec Negotiation with SIP/SIP-I, MSC 13A
› FDS: Handling of SIP Preconditions, MSC 13A
Note: There is no FDS for AMR-WB !

Function Specifications:
› TrFO Interworking with SIP and SIP-I
› Support of AMR-WB Speech Codec
› Out of Band Transcoder Control in MSC Server, GMSC Server and TSC Server
› Media Gateway Selection in MSC Server, GMSC Server and TSC Server
› Support of G729 Speech Codec (not yet released)

User Guides:
› User Guide for Interworking between MSS and External Networks Using SIP or SIP-I (in
preparation)
› User Guide for SIP and SIP with Encapsulated ISUP Signaling, MSC Server (in
preparation)
› MSC Server Inteworking with External Networks Using SIP and SIP with encapsulated
ISUP (not yet released).
› User Guide for Control Setting of AXE Supplier SIP or SIP-I Codec Parameters
Call Flow: Basic SIP and SIP with ISUP Encapsulation (SIP-I) Call Flow (in preparation)
› See also Protocol Specification on FAY level (PU EX SIP and SIP-I) for details on
signaling sequences.
HD Voice | 2012-09-24 | Page 48 (59)
FIP PHASE (TEST)

› Emergency Recovery Procedure (ERP)

› Trouble Shooting Guideline (TSG)
› Data Collection Guideline (DCG)
› External Test Report
› Demo Description
› Feature Configuration Document (FCD)

HD Voice | 2012-09-24 | Page 49 (59)

Configuration
OPTIONS
Overview of AMR-WB
configuration OPTIONS
› New configuration options are introduced on the node level:
– AXE customer parameter to determine whether AMR-WB speech codec is
supported for the interwork towards SIP or SIP-I networks
(AXEMGCFCODECC:ACTIVATION7)
Note: Separate activation for SIP or SIP-I interface through different values is
possible
– AXE customer parameters to determine priority of AMR-WB speech codec
in codec list sent to succeeding node (AXEMGCFCODECC:
PRIORITYSIP7 and PRIORITYSIPI7, applicable only when initial INVITE is
received without SDP offer from preceding node)
– AXE supplier parameters to determine codec used on Nb interlink between
two SIP or SIP-I controlled interfaces, the new subset added for case when
AMR-WB(set0) is used on SIP or SIP-I (CONRMILS:IMSCxXCxILNB)

› New configuration option is introduced on the SIP or SIP-I route level:

- Generic route parameter to disable AMR-WB speech codec towards
particular SIP or SIP-I network (ICP, applicable also to other speech codecs –
see Other Configuration Options).

HD Voice | 2012-09-24 | Page 51 (59)

Overview of PRECONDITIONS
configuration OPTIONS
› New configuration option is introduced on the node level:
– AXE customer parameter to enable preconditions mechanism
(SIPSUPPORTC:PRECONDACT)
› New configuration option is introduced on the SIP or SIP-I route level:
– Generic route parameter to disable preconditions mechanism towards
particular SIP or SIP-I network (PRECOND). Applicable only to outgoing
routes.

HD Voice | 2012-09-24 | Page 52 (59)

OVERVIEW OF OTHER
CONFIGURATION OPTIONS
› Following new configuration options on the SIP or SIP-I route
level are also introduced:
- Generic route parameter to indicate initial direction attribute (IDA) sent
in initial SDP offer from MSC-S. Applicable only to outgoing routes. Also
applicable only in e2e codec negotiation mode or when preconditions
mechanism is used.
- Generic route parameter to disable e2e codec negotiation (i.e. to force
link-by-link codec negotiation) on particular outgoing SIP or SIP-I route
(FLBL)
- Generic route parameter to disable particular speech codecs towards
particular SIP or SIP-I network (ICP). This parameter overrides existing
node configuration (AXEMGCFCODEC:ACTIVATIONx).
Note: This parameter is not dependent to the new “TrFO Interworking with
SIP or SIP-I feature”. Also, there is no configuration option to modify codec
priorities (as received from preceding node) on per route basis

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CONFIGURATION OPTIONS - DT

› See feature FCD for DT example.

› All the feature relevant DT could be also found at:

HD Voice | 2012-09-24 | Page 54 (59)

KNOWN ISSUES AND
LIMITATIONS
Known ISSUES AND
LIMITATIONS (1/2)

› Problems with VoIP clients non-compliant to RFC and 3GPP

26.114 (non-MGTS clients) when interworking with IMS:
- Mode-set or mcc/mcp not indicated by client for AMR multimode
codecs in SDP offer or answer - results with codec removal or call
release
- Clockrate mismatch between AMR-WB codec and telephone event
(or support for telephone-event/16000 not indicated by client) – this is
against RFCs and not-supported by M-MGw, results in problems with
DTMF and TrFO
› These problems can be addressed through Client or SBG
configuration, otherwise through market customizations

HD Voice | 2012-09-24 | Page 56 (59)

Known ISSUES AND
LIMITATIONS (2/2)

› SRVCC Handover – effectively disables both e2e codec

negotiation and SIP Preconditions (still likely to achieve
TrFO through the link-to-link codec negotiation but maybe
not original codec). Current standardization is not
completely clear, so improvements are expected in the
future. It could be considered to use SIP signaling for
negotiation of codec towards IMS (‘Empty’ INVITE) before
actual session is established.

HD Voice | 2012-09-24 | Page 57 (59)

NODE DEPENDENCIES

› M-MGw R6 FP2 with EP8 or a later compatible profile (to

support AMR-WB codec on Mb interface)
› OSS-RC: 13.0.5 is the minimum due to support for the new
statistic counters. Verified with this version.

HD Voice | 2012-09-24 | Page 58 (59)