HD Voice

Interconnection Workshops

PIERRE Marie-Cline Orange Labs Networks

May 2017 v6
 Objectives

1 Prerequisite : Introduction to IP interconnection for

communication services

2 Understand HD Voice

3
Understand constraints and implementations of HD Voice
interconnection

2 interne Orange
 Agenda

part 1 HD Voice : what is it?

part 2 HD at interconnection
Pre requisites
End-to-End codec negotiation
part 3 Media plane recommendations
part 4 Implementation of HD interconnection for Orange

3 interne Orange
 Introduction

44 interne Orange
 >300 millions HD Voice customers worldwide
(source GSA) http://www.gsacom.com/hdvoice/

164 HD Mobile Network Operators

reached in May 2016
Mobile HD deployments mainly over 3G (13
MNOs with 3G & 2G)

including 63 VoLTE deployments

Orange estimated customer base:

>20 millions (Jan. 2015)
In some markets, such as France, over 50%
of the installed mobile customer base now has
HD voice devices

Increase HD Call probability next step is HD Interoperability to join the HD islands

Pre-requisites (IP to IP interco) progressively available, still technical limitations

VoLTE is the new HD Voice frontier

GSMA IR 92: WB-AMR must be supported in devices if HD Voice is offered by operator

http://www.voixhd.orange.com/fr/Accueil
 Current Orange deployments : 22 mobile markets

+ O. CA
04 2015

About 10/12
+ 2Gmillion of HD terminals
HD Voice launched sold
in 2014 by Mobistar and O. Poland

+ Fixed HD Voice : France, Spain, Poland and Tunisia

 HD Voice ingredients: Codec, Devices & Network QoS
Network Codecs Audio bandwidth
G.711, G.729 Narrowband
Fixed
G.722 Wideband (HD Voice)
AMR Narrowband
Mobile
AMR WB Wideband (HD Voice)

HD speech codec

Network QoS
Devices

Devices quality is guaranteed by : Network QoS supervision is mandatory to guarantee

HD Voice Label (GSMA) for Mobile Phone HD Voice Quality:
Acoustical performance Live monitoring of:
Noise Reduction Packet loss
Delay Jitter
End to end delay
CAT-iq certification and GSMA HD voice label Troubleshooting of detected problems
(since June 2013) for fixed (DECT) phone

7 interne Orange
 Overview on Orange mobile HD voice architecture

Pre-requisites
BICC
MSC MSC

R4 MSC, IP backbone

AoIP (if HD over 2G wanted)

RNC Iu CS MGW MGW BSC
(3G) Nb (TrFO) (2G)

WB-AMR codec support needed

(Orange choice for 3G/2G: limited to set 0 (6.6 kbps/8.85 kbps/12.65 kbps))

Mobile 3G HD Voice devices range : >180 HD Voice mobile phones from 13

different suppliers (March 2015)
- Apple, Nokia, Samsung, HTC, Sony, LG, ZTE, Huawei, Motorola, Blackberry,
Alcatel, Intel, Orange
- 2G HD devices from: Nokia, LG, Sony, HTC, Huawei
 Overview on Orange fixe HD voice architecture

CAT-iq 2.0 by DECT Forum & Home Gateway Initiative (HGI) with G.722 codec
Partnership agreed between GSMA / DECT Forum in June 2013 to extend HD Voice
logo to fixe HD devices.
Evolution towards rich services with CAT-iq 3.0

First implementations : USB / Today : Livebox with CAT-iq inside

CAT-iq dongle (Livebox Play et Livebox 3 PRO)

SIP stack

SIP stack
Improved user experience
Ease HD adoption
Latest Orange HD Devices : D47, D49, D68, M55,
M70, Allure
 HD at interconnection
- Pre requisites

10
 Reminder of possible IP interconnections for a DNF

National National
Mobile TPO Fixed TPO
Signalling: SIP
NB codecs: G.711,
Signalling: SIP-I or SIP G.729
NB codecs G.711, AMR WB codec: G.722
CS7, AMR CS1
WB codec: WB AMR IP interconnection
infrastructure
IP interconnection
infrastructure

Signalling: SIP-I
NB codecs G.711 DNF network

Signalling: SIP or SIP-I

NB codecs: G.711, G.729, AMR
TDM
WB codecs: WB-AMR, G.722
interconnection IP interconnection
infrastructure infrastructure

International
TDM based carrier
network
 Mobile-Mobile HD interco (nat, intl)

Prerequisite: IP-IP Interco with SIP(I)* interface including HD voice

1. WB-AMR codec supported on a SIP(I) interface
2. End-to-end codec negotiation across BICC/SIP(I) (TrFO feature on MSC/MGCF) :
to select WB-AMR codec E2E and avoid transcoding.

HD Voice (WB-AMR E2E)

SIP-I
International
OINIS third party
SIP-I NBS
MGCF I mobile
Orange RTP 5200 operator** WB-AMR
S IP
Mobile Core
B interco
network
MGW C RTP SIP-I
WB-AMR I MGCF
Third party
S
mobile
B
RTP operator*
C MGW WB-AMR

* BICC might be used in some cases for national interconnection between mobile operators.
 Fixed-Mobile HD interco (nat)

HD Voice (WB-AMR/G.722 transcoding)

SIP-I SIP
MGCF I SIP-I SIP-I
I CS Third SIP
Orange
S S party fixed
Mobile Core IP interco
B B operator
network
IM-MGW C RTP RTP C RTP IM-MGW RTP G.722
WB-AMR

Transcoding location ?
G.722
WB-AMR

Prerequisite: IP-IP Interco with SIP(I) interface including HD voice codecs

1. End-to-end codec negotiation (TrFO feature)
2. WB-AMR/G.722 transcoding capabilities:
Available on IM-MGW Audiocodes (6.2.107) and I-SBC ACME 6300
Italtel CS : HD transcoding (WB-AMR/G722) in 2nd priority versus G711 (end-to-end codec
negotiation).
Not fully available yet over mobile IM-MGWs (no requirements from operators)
Ericsson: feature planned with MSS15A
Huawei : on ATCA HW with CS9.2 release (not planned on CPCI). Not validated yet.
 HD voice is always better than NB voice
3 HD interconnections are possible:
Mobile to Mobile
Between 2 HD operators, AMR-WB codec will be used

Subjective tests in Orange labs (in 2010) on Mixed (NB-WB) Scale .

Tests according to ITU-T P.800 standard
Fixed to Fixed
Between 2 HD operators, G.722 codec will be used

Mobile to Fixed
Different codecs between fixed and mobile
networkstranscoding between G.722 and AMR-WB
(and vice and versa)
 HD at interconnection
- End-to-End codec negotiation

15
 E2E codec negotiation mechanisms : TrFO
(Transcoder Free Operation)

Introduced with MSC-S R4 to allow End To End (E2E) codec

negotiation in mobile Core Network with BICC CS2 signaling
protocol.
Avoid transcoding (same E2E codec)
Enhance voice quality (negotiate the best codec)
Allow HD voice (E2E HD codec transport)
Allow bandwidth saving (transport of compressed codec)

Codec negotiation leads to TrFO if the same codec and same

codec set is used in the end to end media path
Oobtc mechanism to allow Trfo

Transcoder Free Operation: configuration of a speech or

multimedia call for which no transcoder device is physically
present in the communication path and hence no control or
conversion or other functions can be associated with it
 E2E codec negotiation mechanisms : Oobtc
(Out of Band Trancoder Control)
Oobtc mechanism described in 3GPP TS 23.153 with BICC CS2 to
allow TrFo
OoB Codec O-MSC Transit T-MSC
Control MSC Negotiation MSC
Server Server O-MGW Transit T-MGW
Plane T MGW
r OoB Codec
Codec List (v, w, x, y, z)
OoB Codec RANAP RANAP Negotiation
MGw a MGw
Negotiation n Codec List (v, w, x, z)
Control Control
s Selected Codec = v

Bearer Req i Req

Bearer Bearer Req Selected Codec = v, Available List (v, x, z, )
ME ME
RNC MGW t MGW RNC Selected Codec = v
Selected Codec = v, Available
N List (v, x, z, )
User e
t Selected Codec = v
Plane
w Bearer Established Bearer Established
o
Radio Bearer Iu Bearer CN bearer Iu Bearer Radio Bearer
r
k

End to end connection

Extended to SIP-I in 3GPP TS 23.153 release 8 => introduction 3GPP

Oobtc indicator in the SDP

Available Codec List transmitted to the Originating node can be used

in Mid Call Codec Negotiation (MCCN) (Re-negotiation BICC (APM) of
the Selected Codec and Available Codec list during the call , e.g.
during HO)
Extract from Orange Labs HD Interco status for IBNF
 E2E codec negotiation : basic principles

Codec List (offered) Codec List (offered)

1- WB AMR CS0 1- WB AMR CS0
2- AMR CS7 2- AMR CS7
3- ,,, 3- G711

1 - codec offer
SIP or SIP-I
GMSC /
BICC MGCF
2 - codec offer
Preceding Network Succeeding Network

3 - codec answer
4 - codec answer
Iuup
MGW RTP

Codec answer (and Codec answer (and

used on media plane) used on media plane)
WB AMR CS0 WB AMR CS0

HD (WB AMR) E2E

HD Voice can be negotiated end-to-end

 Main available standards for interworking
interface
3GPP TS 29.165

3GPP standardisation bodies:

IMS SIP ITU-T
No standard for interface
3GPP (TS)
interworking
3GPP TS
24.229
between ITU-T
SIP-I and SIP TS 29.235
available
TS 29.163
3GPP SIP-I
ITU-T
(Nc)
Q1912.5
Profile C TS 29.235
interface interface
TS 29.235 3GPP TS
29.231
ISUP/
ITU-T SIP-I
TS 29.164 BICC
SIP interface between two IMS networks is defined in 3GPP TS 29.165
No standard for interworking between ITU-T SIP-I and SIP available
TS 29.235, TS 29.163, TS 29.165 - networks externally compliant with IMS are also implicitly
covered
TS 29.164 and TS 29.235 - define interworking with external SIP-I Q1912.5 profile C
 Interworking between CS network and external IMS
network (based on SIP or SIP-I) : complexity of standards
leading to different implementation by suppliers

TS 29.163 chap 7.3

BICC : TS 23.205 SIP
TS 29.235 chap 7
TS 29.164 chap 6.3
OR OR

TS 29.235 chap 4
SIP-I: TS 23.231 SIP-I

CS network MGCF External IMS Network

doc OLPS/COMSERV/SVQ/ISI

It describes :
the codec negotiation and parameters mapping at MGCF
the packet adaptation at IM-MGW level, for instance between IuUP
packets of CS/BICC side and RTP packets of SIP side
CS network and external IMS network (based on SIP or SIP-
I), one example : interconnection between Orange mobile
members and a third party mobile operator or IBNF

I
BICC : TS 23.205 TS 29.164 chap 6.3 SIP-I
S
CS Orange MGCF B Interco with mobile
Members Orange Members
C third party or OINIS
(SONUS)

SIP-I based TS 29.164 => OoBTC is optionnal

When the IWU sends a SIP request with an SDP offer towards the external SIP-I based network, the IWU
may include the 3GPP_OoBTC_Indicator in accordance to subclause 6.6.3 of 3GPP TS 29.235

Whereas with SIP-I on Nc (TS 29.231) or SIP-I based TS 29.235 =>

OoBTC is mandatory
The IWU shall follow the procedures defined for a 3GPP Intermediate Node in clause 9 of 3GPP TS
23.153
E2E codec negotiation : focus on TrFo for SIP(I)
required on mobile MSC (MGCF)
The goal of the function TrFO for SIP(-I) is to enable an end-to-end
codec negotiation over SIP(I).
This end-to-end codec negotiation will provide:
Optimal audio quality (allows HD Voice and improves also narrow-band
quality)
Saving of transcoding resources.
Optimal Core Network Bandwidth management.

Same license will provide same kind of benefits for several use cases :
Interconnection to other operators (mobile or fixed)
Interconnection to Voicemail
VoLTE :
HD negotiation for VoLTE to/from 3G calls
Improved SRVCC (VoLTE to 3G/2G mobility) management (Quality
and Media Handling).
 E2E codec negotiation : improves audio quality and
transcoding usage but required careful E2E consistency
Ordered codec list Ordered codec list
intra PLMN configuration defined for Interco defined for Interco
3G configuration: link: link:
HD = UMTS AMR WB CS0 1- WB AMR CS0 1- G722
2- AMR CS7 2-G729 intra VoIP configuration
SD = UMTS AMR 2 CS1 & CS7, G722, G711 & G729
UMTS AMR CS1 & CS7 3- AMR CS1 3-PCMA
2G configuration AoIP & AoTDM 4- PCMA 4-PCMU
SD = FR AMR CS1, HR AMR CS1, + 5- PCMU
GSM codecs
HD = 2G AMR WB CS0

BICC GMSC SIP or SIP-I SIP SIP

MSC IBCF I-CSCF
Mobile Core
DNF Fixe VoIP Core
network Net network
RTP RTP
MGW MGW IBGF MGW
Iuup RTP

What are the impacts of codec type/set heterogeneity?

To use transcoding resources in spite of the fact that it is now possible to avoid them.
To lead to non optimal audio quality
To use potential extra bandwidth compared to optimal situation (if G.711 is selected for example)
Note that, if G.711 is selected for the interconnection link, no regression expected for transcoding
ressources and audio quality compared to current TDM situation
But no improvement
E2E codec negotiation : improves audio quality and
transcoding usage but required careful E2E consistency

Codec List at interconnection shall be set carefully

Main drivers to be considered for codec configuration/negotiation rules

are :
Interconnection technical configuration are defined by a negotiation between
players (operator, ISP and regulator if any)

Avoid transcoding as much as possible :

always propose a SD codec in association with HD codec (G711 is the usual
default codec)

For mobile-fixed HD voice interconnection : AMR-WB/G.722 transcoding

must be chosen prior to G711 E2E (not obvious in suppliers
implementations)

At interconnection, Bandwidth is money ! Codec bandwidth consumption:

~100 Kbits/s for G.711, ~40 for G.729A, ~30 for AMR/AMR-WB
 Media plan
Recommendations

25
Codecs over SIP/RTP networks

The SIP SDP parameters, related to codec type and configuration are
described in :
RFC 4867 for AMR & AMR-WB codecs
the payload format supports interoperation with existing transport
formats of AMR and AMR-WB on non-IP networks
RFC 3550, 3551 for support of G.711
RFC 3550, 3551 and 5993 for support of GSM FR, EFR, and HR

The recommendations on these parameters are defined in TS 26.114 &

29.163. They also defines how to perform the mapping interworking
between BICC parameters and SIP SDP parameters.
 Media plan Recommendations
Recommendations
M1 The minimum set recommended for codec list in the INVITE messages sent from mobile DNF towards IBNF, is
G.711 codec with a packetisation time equal to 20ms.

M2 For DNFs supporting HD voice on 3G/2G mobile network, the recommended codec is AMR-WB in bandwidth
efficient , conf 0. WB AMR set 0 (6.6 kbps, 8.85 kbps, 12.65 kbps)* ;
Payload type = dynamic between 96 and 127
octet-align = 0 (bandwidth-efficient operation) ; channels = 1
Media format specific parameters mode-set=0,1,2
Media format specific parameters mode-change-period=2
Media format specific parameters mode-change-capacity=2
Media format specific parameters mode-change-neighbor=1
Media format specific parameters max-red=0
The requirements for WB-AMR are taken from 3GPP TS 26.103 3rd Generation Partnership Project ; Technical
Specification Group Services and System Aspects;Speech codec list for GSM and UMTS
*For VoLTE : Full range, up to 23,85kpbs
G722 is the codec used for HD voice in fixed VoIP services. G722, Ptime=20 ms ; (Payload Type static =9)
Rf Rec. ITU-T G.722, 7 kHz audio-coding within 64 kbit/s, Nov. 1988

M3 Fax modem calls are supported by default by using the G.711 A Law codec without media session
modification. NOTE This means that fax modem calls must be established with G.711 A Law as the initial
negotiated codec.
Alternatively, T.38 mode or Clearmode codec can be used if agreed by both connecting parties. V.152 is
optional
M4 For Modem, 64 kbit/s transparent calls. When the encapsulated ISUP Transmission Medium Requirement
parameter is set to 64 kbit/s unrestricted, the SDP contains Clearmode codec [RFC4040] as described in
table 6 of ITU-T Q.1912.5.
M5 The method recommended for DTMF transport is Telephone Event (RFC 4733).
Note that if AMR-WB is present in SDP Offer, telephone-event/16000 must be proposed in addition to
Telephone-event/8000
27 interne Orange
 M5. Recommendations on DTMF based on RFC
4733 with HD Voice in SIP/SIP-I
DTMF clock rate in SDP:
RFC 4733 requires that telephone events and voice codecs use the same clock rate in a RTP
media session (clarified by RFC 4733 errata Errata ID 3489).

Application to HD Voice
SDP Offer/Answer must contain telephone-event/16000 if a Wideband audio codec is
proposed/negotiated (except G.722 which use a clock rate at 8kHz).
m=audio 49152 RTP/AVPF 97 99
a=rtpmap:97 AMR-WB/16000/1
a=fmtp:97 mode-change-capability=2; max-red=220
a=rtpmap:99 telephone-event/16000/1 AMR-WB codec and DTMF RFC 4733 mode
a=fmtp:99 0-15

SDP Offer must contain telephone-event/8000 and telephone-event/16000 respectively

if Narrowband and WB-AMR audio codecs are proposed.
m=audio 49152 RTP/AVPF 97 98 99 100 101 102
m=audio 49152 RTP/AVPF 97 98 99 a=rtpmap:97 AMR-WB/16000/1
a=rtpmap:97 AMR/8000/1 a=fmtp:97 mode-change-capability=2; max-red=220
a=fmtp:97 mode-change-capability=2; max-red=220 a=rtpmap:98 AMR-WB/16000/1
a=rtpmap:98 AMR/8000/1 a=fmtp:98 mode-change-capability=2; max-red=220; octet-
a=fmtp:98 mode-change-capability=2; max-red=220; octet-align=1 align=1
a=rtpmap:99 telephone-event/8000/1 a=rtpmap:99 telephone-event/16000/1
a=fmtp:99 0-15 a=fmtp:99 0-15
a=ptime:20 a=rtpmap:100 AMR/8000/1
a=maxptime:240 a=fmtp:100 mode-change-capability=2; max-red=220
a=sendrecv a=rtpmap:101 AMR/8000/1
a=fmtp:101 mode-change-capability=2; max-red=220; octet-
2 x AMR narrowband codecs and align=1
DTMF RFC 4733 a=rtpmap:102 telephone-event/8000/1
Both narrowband and wideband codecs
Extract from IMT/OLN/CNC doc DTMF based on RFC 4733 with HD voice in SIP/SIP-I and DTMF RFC 4733
 Implementation of HD
interconnection

29
 HD interco roll-out : issues still to be solved

Due to ambiguity in standards (see previous slides), different

implementations and behaviors are encountered on our suppliers
equipment :
Handling of DTMF 16k (clarified with RFC 4733 errata [Errata ID 3489]
see slide 21)
TrFo implementation over BICC / SIP-I : different choices of
implementation for codec negotiation (with/without preconditions,
OoBTC, linked to TS 29.235 or TS 29.164)

These different behaviors are not always compatible, and may lead
to the impossibility to get HD voice end-to-end negotiation in some
cases
See following slides

30 interne Orange
 mobile HD interco : summary of current status (May 2016)

MSC Ericsson No issue (validation done with MSS 13A)

Licence Trfo SIP/SIP-I :
Oobtc not supported
End2end codec negotiation (IAM -> INV -> IAM) without ACL sent to the caller CN
Precondition ON/OFF
Beware : G.711 Transcoding on the MGW E/// if absence of DTMF 16k, leading to loss of HD quality

MSC Huawei Implementation :

Different features/licences for SIP_i interworking : SIP-I on Nc or SIP/SIP-I interworking
Oobtc legal only with the licence SIP-I on Nc (additional limited cost if the O. member already use the SIP/SIP-I interworking one)

Status :
TrFO over SIP-I between Huawei and Sonus is now globally working fine on the top of CS10.2 (MSOFTX3000 CS10.2 SPH116+ UMG8900
CS10.2 SPC121 patches)
However, 2 not-fully optimized results leading to the creation of 2 CRs (Change-Request) to Huawei:
Unexpected Need of FPTC on originating MGW
Unexpected Usage of extra-TC CORE

MSC NSN Implementation (partial tests done with MD16,1): NSN requires OoBTC and/or Preconditions ON, or additional License 3178 for E2E codec
negotiation
TrFo possible in SIP-I
by activation of Oobtc or licence 3178 implying MCCN behaviour
by activation of preconditions to perform end to end codec negotiation (IAM -> INV -> IAM)

Issues :
Potential issues unknown (e.g DTMF 16k,) : not validated

Sonus NBS Issues :

No support of DTMF 16k (=> G.711 Transcoding on the MGW E///) : Correction wit release 4.2. Basic case ok. To be checked in more
complexed ones (OK with Huawei MSC with not-fully optimizd behaviors)
Oobtc managed by SMM (INVITE and 18x). Does not work in the SIP-I subsequent messages (e.g. UPDATE)
Wrong management of the HD transcoding (AMR-WB G.722) for mobile to fixed calls => Still no satisfying corrections
Wrong management of the precondition => Sonus not fully transparent

ISBC Oracle ISBC 9200 bad interworking management between DTMF 16K &8K:
in Mobile-Fixe use case (fix ISBC only handles DTMF 8k in SDP, leading to loss of HD with E// MSC for instance) : ISBC 9200 will not
be fixed (end of life). OK on new 6300 platform (fixed in 7.2).
in Mobile-Mobile use case : Payload type mismatch when 3GWB to 3GNB calls : fixed from 7.2.3F3P6

CS Italtel HD transcoding (WB-AMR/G722) in 2nd priority versus G711 (end-to-end codec negotiation). No modification planned on CS. Should be ok on
31 interne Orange
Italtel NBI.
 E2E overview : mobile-mobile interco - Double
OK? TC

(Orange footprint) To be
on bothtested
side
with high
impact on
HD quality

Huawei MSC
withCS10.2
with CS9.2 +
&
SPH116
CS10.2 without
SPHxxx
Ok in theory
Not validated
since
corrections
SIP-I

-TrFO NSN
E/// depends on
MSC precondition or
OK from OoBTC indicator
MSS13A : DTMF 16k not
tested
-TrFO NSN
-
depend son
precondition or
SIP-I OoBTC indicator :
OoBTC ind. not
implementet dby
E//
-DTMF 16k not NSN MSC
tested MD16.1
-

?
 E2E overview : International interco
France (Orange footprint)
IBNF
fixed VoIP - codec OK with
T3G SIP list issue SIP Sonus CS10.2
for WB to NBS SPH116
G722 with 4.2 (with some Huawei MSC
calls SW unoptimized
SIP-I
behavior)

SIP-I
SIP-I

-TrFO NSN depend

on
OK precondition/OoBT
C indicator DTMF
16k not tested
-

E///
MSC NSN MSC

Restricted
 Next steps for HD
interconnection

34
 Super HD : Enhanced Voice Service (EVS)
EVS-FB

Superior quality
EVS-SWB
EVS-WB and EVS AMR-WB IO
Super HD quality (starting at the same bit EVS-NB

rate as HD voice - around 12-13 kbit/s) Fullband

Super-wideband
Better NB and WB quality at same rate Wideband

Good music quality at low delay (16.4 kbit/s Narrowband Frequency

in SWB) (telephone-band) (Hz)

EVS part of standardized e2e voice service 50 100->300 3400->4000 7000->8000 14000->16000 20000

Interoperability AMR
AMR-WB
Intrinsic interoperability with HD voice
Rel-12 phone
(improved AMR-WB inside EVS)
Standardized codec with worldwide support EVS
Efficiency (capacity, coverage) AMR-WB IO

EVS bit rates optimized for LTE TBS; wide (Enh.) AMR-WB
range of bit rates to cover also fixed-line
AMR
applications
Better robustness against packet losses
Jitter Buffer Management included (only 5.9 7.2 8 9.6 13.2 16.4 24.4
recommended) 32 48 64 96 128
Current NB/WB quality at a lower bit rate

35 interne Orange from study by

OLPS/COMSERV/SVQ/
 Super HD (EVS) : Network impact
key impacts on terminals + IMS (P-CSCF) + Legal Interception

Interco between Operators : full SIP interco needed for super HD

Oracle NBI : support in passthrough mode. Transcoding planned for Cz800 release
Sonus NBI : support in passthrough mode. Transcoding capability may come 2nd
36 interne Orange
half 2017 (release 6.0 ?) from study by
OLPS/COMSERV/SVQ/
 Conclusion

37
 To Sum-up, what is required

1. IP interconnection (with SIP/SIP-I control plane)

2. HD interco features :
End to end codec negotiation (TrFO feature on MSC)
End to end support of HD codec
WB-AMR for mobile
+ WBAMR/G722 transcoding for fixe/mobile
interconnection

3. Solve the remaining issues to enable roll-outs

Suppliers implementation of fixes for remaining bugs
Define best configuration allowing and optimizing end-to-end
HD voice quality
Tests E2E configuration according to affiliates needs

38 interne Orange
 Contact points

Questions, comments?
Or you would like to give us a feedback on IP interconnection in
your DNF?
Please contact us at:

Interco.FrontOffice@orange.com
 References

[1] Recommendations from SIP TG "DTMF based on RFC 4733

with HD voice in SIP/SIP-I
[2] WB-AMR recommendation for 3G affiliates_October 2010-
v1.1.doc
[3] Interco training : Introduction to IP Interconnection for
communication services
[4] Interco training : Interconnection of mobile DNFs with iBNF
[5] NetComm : Impact study of the EVS codec in VxIMS
Thank You
Annexes
 Glossary

IPX IP exchange
ACL Available Codec List
I-SBC Interconnection SBC
BGW Border GateWay
IWU InterWorking Unit
CDR Call Detail Record
MCCN Mid Call Codec Negotiation
CS Circuit Switched
MGW Media GateWay
CSCF Call Session Control Function
MGCF Media Gateway Control Function
DNF Domestic Network Factory
MNO Mobile Network operator
DNS Domain Name System
NAPT Network Address & Port Translation
E2E End-to-End
NP Number Portability
GRX GPRS Roaming eXchange
OoBTC Out of Band Transcoder Control
GW GateWay
PLMN Public Land Mobile Network
IBCF Interconnection Border Control Function
PS Packet Switched
IBNF International & Backbone Networks Facory
PSTN Public Switched Telephone Network
I-BGF Interconnection Border Gateway Function
SIP Session Initiation Protocol
IC International Carriers
SIP-I SIP with encapsulated ISUP
II NNI Inter-IMS Network-to-Network Interface
TDM Time Division Multiplexing
IM MGW IP Multimedia Media Gateway
TrFO Transcoder Free Operation
IMS IP Multimedia core network Subsystem
TrGW Transition Gateway
TPO Third Party Operator
 DTMF clock rate interworking issue: example
between Ericsson MSS and Acme I-SBC
Ericsson
MSS13A SDP offer:
AMR-WB/16k
Acme I-SBC
DTMF/16k

MSC SIP / SIP-I

SDP answer:
AMR-WB/16k
Mobile Core DTMF/8k

AMR-WB AMR-WB
DTMF "in- DTMF RFC
band audio" 4733 8K
MGW
PCMA
transcoding AMR-WB/16k and DTMF/16k are proposed in the SDP offer (INVITE sent by MSC
E///).
AMR-WB/16k and DTMF/8K are selected in the SDP answer sent by I-SBC (200 OK).
Voice and DTMF clock rates mismatch. MSC requests to the MGW:
- Voice codec = AMR-WB/16k
No End to End HD Voice! - DTMF mode = "in-band audio"

No DTMF integrity! AMR-WB is used for voice between I-SBC and MGW but internal PCMA transcoding
is performed in the MGW (due to DTMF "in-band audio" activation).
MGW sends DTMF "in-band audio".
SBC sends DTMF RFC 4733 8k.
Extract from IMT/OLN doc DTMF based on RFC 4733 with HD voice in SIP/SIP-I
Managed VoLTE/VilTe
VoLTE/vilTE media handling

3GPP Rel 12 will provide a new release of MTSI/VoLTE specifications for enhanced
media handling : enhanced Bandwidth & QoS management for VILTE, super HD
VoLTE with EVS and fixed mobile interop.

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Signalling IOT between BICC and SIP/SIP-I networks

Codec parameters for 3GPP AMR-WB codecs (3GPP TS 29.163 B.2.5.2)

Single Codec information element SDP payload format parameters
Codec IDentification Config- Payload Encoding Other Parameters
WB- Type name (NOTE 1)
Code number
FR_AMR-WB or 0 dynamic AMR-WB mode-set=0,1,2
OHR_AMR-WB
OFR_AMR-WB or 0 dynamic AMR-WB mode-set=0,1,2
UMTS_AMR-WB (NOTE 2)
OFR_AMR-WB or 1 dynamic AMR-WB mode-set=0,1,2
UMTS_AMR-WB dynamic AMR-WB mode-set=0,1,2,8
dynamic AMR-WB mode-set=0,1,2,4
(NOTE 2)
OFR_AMR-WB or 2 dynamic AMR-WB mode-set=0,1,2,4
UMTS_AMR-WB (NOTE 2)
OFR_AMR-WB or 3 dynamic AMR-WB mode-set=0,1,2,4
UMTS_AMR-WB dynamic AMR-WB mode-set=0,1,2,8
dynamic AMR-WB mode-set=0,1,2
(NOTE 2)
OFR_AMR-WB or 4 dynamic AMR-WB mode-set=0,1,2,8
UMTS_AMR-WB (NOTE 2)
OFR_AMR-WB or 5 dynamic AMR-WB mode-set=0,1,2,8
UMTS_AMR-WB dynamic AMR-WB mode-set=0,1,2,4
dynamic AMR-WB mode-set=0,1,2
(NOTE 2)
NOTE 1: Payload types for FR_AMR-WB, OHR_AMR-WB and OFR_AMR-WB shall include the mode-change-
period=2 parameter and should include the mode-change-neighbor=1 parameter.
NOTE 2: Payload types for UMTS_AMR-WB should include the mode-change-period=2 and mode-change-
neighbor=1 parameters, normally used for signalling GSM AMR-WB codecs, to assure end-to-end
interoperability with OoBTC and TFO. Its actual capabilities would otherwise be signalled without these two
parameters.

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SIP/SIP-I SDP for AMR-WB
3GPP TS 26 236 & 29.163 defines the values to be applied
for the AMR-WB Media Type Registration defined in RFC 4867 Chapter 8.2

name AMR-WB Proposed value

Octet Aligned =0: bandwidth efficient
Can be both on SIP side
=1: octet aligned
Mode-set (table 1a) 0,1,2,3,4,5,6,7 (all codecs) See Operator Mobile network
configuration
Mode-change-period =1: (every 20ms)
=2
=2: (every 40 ms)
Mode-change-capability =1: clients non capable of restricting the mode change period to 2
=2: client capable of restricting the mode change period to 2 =2
Mode-change-neighbor =0: change to any supported modes are allowed =1 // mandatory on 2G to switch to
=1: sended should only perform change to the neighbor mode. neighbour mode. Free on 3G.
maxptime Max allowed is 80 for non redundant. Better to have 20 (1 packet)
CRC =0 : no CRC =0
=1: CRC used and automatically Octet align shall be used (0) (available only in octet-aligned mode)
Robust sorting =0: simple payload used (0) =0
(available only in octet-aligned mode)
interleaving =0 interleaving shall not be used. If interleaving present it is =0
automatically Octet Align(0) (available only in octet-aligned mode)
channels (1) For mono
=1
(2) for Stereo
Max-red (no value) =0 so no redundancy

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