8 the path to voice over lte - vo lte

© 2012 Agilent Technologies

Wireless Communications

1 © 2013 Agilent Technologies


Greater insight. Greater confidence.

Accelerate next-generation wireless.

The Path to Voice over LTE - VoLTE

with IMS (IP Multimedia Subsystem)

presented by

Andjela Ilic-Savoia

Agilent Technologies







Agenda:

• Intro: a very brief history of IMS, Legacy (CS) vs. LTE (PS) domain

considerations

• Architecture of IMS, components, SIP

• IMS call: SIP call-flow

• What happens when out-of-coverage: flavors of fallbacks

• Protocol features for successful and efficient VoLTE

• Discussion, questions, comments?







Agenda:


considerations












IMS - IP Multimedia Subsystem

…how it all began…

• 3GPP defined IMS in 1999

• IMS as the framework for delivery of multimedia services was standardized

in 3GPP rel.5 for delivery of “Internet” services on GPRS. This was updated

and extended to CDMA and WLAN.

• In 2009 a group of over 40 organisations, operators, vendors etc. came

together to form One Voice, whose aim was essentially to decide or drive

the method of voice delivery on LTE through IMS.

• GSMA finally adopted VoLTE in 2010, and so did many of the industry big

companies

• The use of IMS through the Verizon LTE network in the USA has

accelerated the use of IMS development for mobile devices.







Support for Voice with LTE

2G/3G – Circuit Switched calls have an allocated resource even during times of inactivity - even when nothing is being said

• Inefficient use of available bandwidth

• Access times between requesting resource and being able to talk were too slow to enable a reaction based allocation reduced flexibility for resource allocation

LTE – UE will generally only be provided resources when it is necessary – even for voice

• Allows efficient use of network resources. If we are saying nothing we will require no network resources

• Places stress on the network to ensure suitable access timing and quality of service (QoS).

LTE transportation is fully IP – no circuit switched services







Voice with LTE: what is the goal and

how to get there

To deliver same standard of voice call with VoLTE as is delivered by 2G/3G.

Most agree that the long term solution for voice is to use VoIP and an IMS based core network - However it will take time for networks to support this.

For networks which do not support IMS several technologies are considered, namely:

• CSFB (Circuit Switched Fall Back) - single radio approach

• SVLTE or Dual Standby approach (Simultaneous Voice and Data LTE) - dual radio approach

• SRVCC (Single Radio Voice Call Continuity) – Voice on LTE with CS backup

CSFB, SVLTE and SRVCC all involve some level of I-RAT behavior







Enter IMS …

Characteristic CS Mobile

Telephony

Legacy IP IMS

IP Multi-Media

Service

Standards bodies 3GPP, 3GPP2 IEEE 3GPP/3GPP2 and

IEEE

Primary service CS voice PS data PS data and PS voice

Addressing

technique

Phone numbers IP addresses IP + support for legacy

phone numbers

Access Protocol Reservation based Reservation less SIP provides

reservation based

protocol for voice and

video

Connection Type Connection orientated Connectionless SIP provides

connection orientated

Architecture Centralised/ hierarchical/

closed

Distributed/ flat/open IMS “walled garden”

debate

QoS Guaranteed bandwidth Best Effort GBR and BE

supported

AAA

No. of devices ~5 billion >500M Will be lots!







Agenda:


considerations












IMS has quite complex architecture…

Architecture divided into:

• Application Plane, Control Plane, User Plane

Important part of Control Plane is the 1st point of contact for the UE –

The CSCF (Call Session Control Function).

CSCF is further divided into nodes:

• Proxy CSCF (P-CSCF) (acts as the entry point in the IMS core network)

• Interrogating CSCF (I-CSCF)

• Serving CSCF (S-CSCF)







P-CSCF:

Proxy-Call Session Control Function of IMS

• It is assigned to an IMS-capable UE terminal before registration (through OTA message), and does not change for the duration of the registration.

• It may be in the home domain or in the visited domain.

• It facilitates the routing path for mobile originated or mobile terminated session requests.

• It is responsible for allocating resources for the media flows (bandwidth management)

• It can also compress/decompress SIP messages using SigComp, which reduces the RTT over slow radio links.

• It provides subscriber authentication, and is responsible for the security of the messages between the network and the user (ex: may establish an Ipsec).

• It may include a Policy Decision Function (PDF), which authorizes media plane resources e.g. QoS over the media plane.







SIP and IMS

• SIP (Session Initiation Protocol) was initially designed to work in an open

homogeneous IP network

• SIP provides the signalling required to support call set-up procedures

• SIP also provides many other services (caller id, multi-party & emergency

calls…)

PSTN IP Network

GSM

3G IP Network

Telephone Network

SIP server







To test VoLTE today, you need:

• IMS/VoLTE capable device

• IMS/VoLTE capable BSE

• IMS Server

• Somebody to talk to: Either

another UE or IMS Client

emulator







Agenda:

• Presentation:


considerations












IMS uses SIP

(Session Initiation Protocol)

REGISTER REGISTER

OK OK

INVITE

INVITE TRYING

TRYING

RING RING

OK OK

ACK ACK

BYE BYE

ACK ACK

RTP/RTCP

User A User B CSCF Proxy

m=audio 49120 RTP/AVP 98 97

a=rtpmap:98 AMR/8000

a=fmtp:98 mode-set=7

Optional AKAv2 authentication and IPsec

SIP (Session Initiation Protocol) e.g. INVITE, TRYING, RING, OK, BYE etc.

SDP (Session Description Protocol) e.g. m (media), a (attribute) etc.

RTP (Real-time Transport Protocol) e.g. AMR encoded speech

RTCP (RT Control Protocol) e.g. Send/receive quality metrics

CSCF = Call Session Control Function







IMS uses SIP (Session Initiation Protocol)

REGISTER REGISTER

OK OK

INVITE

INVITE TRYING

TRYING

RING RING

OK OK

ACK ACK

BYE BYE

ACK ACK

RTP/RTCP

User A User B CSCF Proxy

CSCF = Call Session Control Function

CSCF Proxy

User A User B IP

Router/IP

network

SIP messages go via proxy server

RTP voice traffic ~ peer to peer (or via

other network nodes)

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RTP

RTP is used for the delivery of the user data,







Agenda:


considerations












What if…LTE/IMS out of coverage?

Then:

the Voice call to work would require some level or I-RAT.

Let’s examine each of the voice related I-RAT behaviors separately.







GSM/W-CDMA/

TDSCDMA

1xRTT/eVDO/

eHRPD

Phone x Network Configuration

SVLTE Simultaneous Voice & LTE aka Dual Standby

•Two phones in one case

•1xRTT (or GSM/W-CDMA) chipset for all voice calls (CS only)

•LTE/eVDO/eHRPD (and/or W-CDMA) separate chipset for data

LTE/eVDO/HRPD radio

1xRTT CS radio CS Voice Client

e.g.

GUI

LTE

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http://www.google.co.uk/imgres?imgurl=http://www.4gamericas.org/UserFiles/image/Board_of_Governors_Logos/LTE-Logo.jpg&imgrefurl=http://www.4gamericas.org/index.cfm?fuseaction=page&sectionid=249&usg=__g9OzpSg8L-C0i9paCUWSl7FbpLM=&h=1200&w=1315&sz=62&hl=en&start=1&zoom=1&tbnid=9Hz9uaH9RwbuPM:&tbnh=137&tbnw=150&ei=g4ekTrW5DM_dsgbUsJyTAw&prev=/search?q=LTE&hl=en&gbv=2&tbm=isch&itbs=1







GSM/W-CDMA/

TDSCDMA

1xRTT/eVDO/

eHRPD


CSFB Circuit Switched Fall Back

•Handover from LTE to legacy 2G/3G for ALL voice calls

•Use Circuit switched voice (and if available parallel slower legacy data)

LTE/W-CDMA/GSM radio

CS Voice Client

e.g.

GUI

LTE










GSM/W-CDMA/

TDSCDMA

1xRTT/eVDO/

eHRPD


LTE / IMS Islands

•Differentiated quality/price IMS voice in LTE coverage areas

•End call at LTE edge

•1xRTT for E911 and wide area coverage (CS only)

CS Voice Client GUI

E.g.

IMS Voice Client LTE/W-CDMA/GSM radio

1xRTT CS radio

LTE










GSM/W-CDMA/

TDSCDMA

1xRTT/eVDO/

eHRPD


SRVCC Single Radio Voice Call Continuity

•IMS voice calling in LTE coverage areas

•Quickly handover from LTE to legacy 2G/3G at LTE edge

CS Voice Client GUI

e.g.

IMS Voice Client LTE/W-CDMA/GSM radio

LTE










GSM/W-CDMA/

TDSCDMA

1xRTT/eVDO/

eHRPD

LTE


SVLTE Simultaneous Voice & LTE

Aka Dual Standby

CSFB Circuit Switched

Fall Back

SRVCC Single Radio Voice

Call Continuity

LTE/ IMS

islands

LTE / IMS only

Most Operators will skip some steps

World phones will need to roam with many network configurations

LTE/eVDO/HRPD/GSM/W-

CDMA/ TDSCDMA radio

1xRTT CS radio CS Voice Client GUI

e.g.

IMS Voice

Client










Agenda:


considerations












LTE Protocol features that will affect

how VoLTE performs

• Multiple PDNs

• QoS through QCI - QoS class identifier

• TFT – Traffic flow Template

• SPS – semi-persistent scheduling

• TTI bundling

• RoHC – Robust header compression

• SigComp –SIP Signaling compression

• Ipsec – security tunneling

• …







Protocol Features:

VoIP QoS and Multiple PDN’s

Strict packet delay-based QoS

QoS will vary by allocation, by

application and will be heavily

dependent on system capacity.

UE’s can have multiple data streams,

Multiple PDN’s, Addresses, Port

numbers etc. – All with different

parameters:

•Default DRB or Dedicated DRB

•Guaranteed or non Guaranteed Bit Rate

•Packet delay budget e.g. 50 to 300ms

•Packet Error rate e.g. 10-2 to 10-6







continued…QoS class identifier : QCI







Protocol Features: Traffic Flow Template

o TFT is set of packet filters ~ EPS bearer

o TFT decides on IP traffic priority

o Sort IP packets based on:

•IP protocol; e.g. UDP, TCP

•Port number

•IP address

•Priority

e.g. Browser

Non-GBR, low error

Dedicated

Internet IMS-SIP

Default

GBR, high error

IPv4/6

address 1

IPv6

address 2

Example configuration

e.g. IMS-SIP signalling

e.g. RTP voice stream

TFT

Applic

ations

IP p

ackets

LT

E R

adio

UE RF

IP2

SIP

RTP

IP1

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Protocol Features:

GBR, TFT, QoS, DRB’s signalled to UE

GBR

TFT

Negotiated

QoS

Dedicated bearer, linked to

default bearer







Protocol Features:

Semi-persistent scheduling (SPS)

• Normal LTE operation – each SF is

allocated individually.

• For highly repeatable applications such

as voice this is un-necessary waste of the

DL signalling channel bandwidth.

• In voice we normally only need a few

100’s of bits at regularly spaced intervals.

• SPS is used to tell the UE to use specific

resources for a fixed time period and can

reduce significantly the DL signalling

bandwidth required.

1 2 3 5 6 7 1 3 5 6 7 1 3 5 6 7 1 DL Normal

DL after

SPS is

scheduled

3 5 4 4 4 4 0 0 0 0 6 7 1 3 5 4 0 6 7 1 3 5 4 0 2 2 2 2 2

1 2 3 5 6 7 1 3 5 6 7 1 3 5 6 7 1 3 5 4 4 4 4 0 0 0 0 6 7 1 3 5 4 0 6 7 1 3 5 4 0 2 2 2 2 2

Normal allocation - Each time an allocation is made, PDCCH resources are required

SPS allocation – reduced PDCCH signalling







Protocol Features: TTI bundling

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5

0 1 2 3

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5

0 1 2 3 0 1 2 3

DL: TTI

UL: RV#

NACK NACK

ACK ACK

If at the cell edge, UE can either increase power or increase coding rate to ensure reception at eNB

If the UE cannot increase its power - eNB gets power headroom status reports from each UE, the UE can be instructed to use TTI bundling.

TTI bundling prevents round trip ACK/NACK delays by transmitting all Redundancy versions in successive TTI’s.

DL signalling is reduced (less ACKs/NACKs) and round trip delay is minimized.

Normal transmission with HARQ feedback

delay of 4ms for each 8ms HARQ cycle – up

to 28ms for final ACK.

Bundled TTI HARQ process 0 in 4

consecutive TTI’s (different RV). Total

feedback delay is only 8ms.

6 7 8 9

ACK NACK

0 1 2 3

6 7 8 9 DL: TTI

UL: RV#







OTA Messages example:

SPS and TTI Bundling







Protocol Features:

ROHC - Robust Header Compression

Typical VoIP Header for IP v4/v6 = 40/60 Bytes

With a typical voice rate of 12kbps, uncompressed IPv6 headers represent approximately 60% of the data sent / received

LTE network efficiency very poor without RoHC

Robust Header Compression is therefore required for LTE VoIP.

Compression is over the air interface only i.e. between UE and eNB/NB/BTS

It happens in PDCP, user plane traffic only.







Protocol Features: RoHC continued

Applic

ations

IP p

ackets

LT

E R

adio

UE RF

Payload Header

Payload

Compressor

LTE Radio Layers

Payload Header

Payload

De-

Compressor

LTE Radio Layers

Network

RoHC

Context

RoHC

Context

• Cuts IP overhead e.g. RTP streams for speech;

2:1 for IPv4, 3:1 for IPv6

• RoHCv2 simplification robustness handling of out-

of-sequence packets

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Protocol Features:

SigComp - SIP Signalling Compression

State full Compressor

/De-compressor

SIP/SDP signalling

stack

RTP/RTCP

Media stack

• Compresses text based SIP and SDP

messages

• Up to 3:1 compression

• Standardized by the IETF RoHC working

group

• only implemented between a UE and

P-CSCF







Protocol Features: IP Sec

Tunnel Mode (VPN like)

• Entire original IP datagram encrypted

Transport Mode

• AH added to protect against alteration of datagrams while in transit

IP Header Data

IP Header AH Data

New IP Hdr AH IP Header Data

Transport Mode

Tunnel Mode

Normal IP traffic

More info in

RFC4301, 4302, 4303







Useful references

GSMA IR.92 IMS Profile for Voice and SMS

3GPP TS 34.229 IP Multimedia call control protocol based on SIP and SDP, UE conformance specification

3GPP TS 33.178 Security Aspects of early IP Multimedia Subsystems (IMS)

3GPP TS 26.114 IP Multimedia Subsystems (IMS) Multimedia telephone: Media handling and interaction

3GPP TS 26.132 Speech and video telephony terminal acoustic test specification

3GPP TS 22.173 IP Multimedia Core Network Subsystem Multimedia Telephony Service and supplementary services

3GPP TS 23.228 IP Multimedia Subsystem (IMS) Stage 2

ITU-T P.862. Perceptual evaluation of speech quality (PESQ)

ITU-T P.863. Perceptual Objecting Listing Quality Assessment (POLQA)

PXT Website. www.agilent.com/find/PXT

Agilent IMS/SIP www.agilent.com/find/E6966A

Interactive Functional Test (IFT). www.agilent.com/find/IFT

http://www.agilent.com/find/PXT

http://www.agilent.com/find/E6966A

http://www.agilent.com/find/IFT







Thank you!

Questions, comments?

Thank you for attending:

The Path to Voice over LTE - VoLTE

with IMS (IP Multimedia Subsystem)

By

Andjela Ilic-Savoia

Agilent Technologies







EXTRAs:







CS vs. PS

RF

RF

RF

IP

IP

Audio

IMS core

emulation

IMS end-point

emulation

Audio

LTE PS

2G/3G CS

2G/3G PS







Agilent E6966A 1FP & 2FP IMS-SIP client

• Easy to install and use Windows 7 PC-based IMS-SIP client

• IPv4, IPv6, Voice, video, SMS, GUI

• Audio: AMR, AMR-WB, G711 a/uLaw, G722/.1, G729, GSM, iLBC, Speex/wb voice codecs

• AMR/AMR-WB Octet-align, bandwidth-efficient, Mode-set negotiation and fixing

• Video: H264, H263

• Play test files, audio loopback, auto-answer

• Proven voice interop with VoLTE UEs







Simple VoLTE Setup







Audio test scenario Human jury testing and/or PESQ with IP impairments

RF RF

Ethernet

Delay/Jitter/Loss insertion

Agilent IMS-SIP server

Agilent PXT Agilent PXT

VoLTE UE VoLTE UE

Audio in/out headphone jack Audio in/out headphone jack

Audio Analyzer







Functional test scenario IMS/CS voice calling & Inter-RAT scenarios

RF

Agilent 8960

1xRTT cell

Agilent 8960

eHRPD cell

Agilent PXT

LTE cell

Ethernet

Test

automation Agilent IMS server and

several remote clients







Delay and Jitter insertion with ZTI NetDisturb

Agilent PXT

VoLTE UE

Windows XP or 7 PC Ethernet

RF

ZTI NetDisturb

USB

USB-Ethernet adaptors can be used with ZTI

NetDisturb to add delay/Jitter to any Ethernet link in

the system. NetDisturb can run on a separate PC or

share a PC with Agilent IMS-SIP client and/or server

Delay/Jitter

insertion

http://www.zti-telecom.com/EN/NetDisturb.html

Agilent IMS-SIP server & client(s)

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Agilent-B&K VoLTE Audio Test System

Agilent PXT

VoLTE UE

Agilent IMS-SIP server & client

Windows 7 PC B&K ZE 0948

Ethernet

USB

RF

Audio Line

in/out

HATS

B&K PESQ Audio Analyzer

.bat file, IPv6

router adv*

* Batch file can be used to configure Windows 7 PC

to transmit IPv6 Router Advertisements. Alternatively

an IPv6 router can be added to the connection

between a Windows 7 or XP PC and the PXT.

B&K Audio Test Software







Adding Jitter to audio quality systems

Agilent PXT

VoLTE UE

Agilent IMS-SIP server & client

Windows 7 PC B&K ZE 0948

Ethernet

USB

RF

Audio Line

in/out

HATS

B&K PESQ Audio Analyzer

.bat file, IPv6

router adv*

B&K Audio Test Software

Optional

Delay/Jitter added

to any link

Delay/Jitter tool,

e.g. ZTI NetDisturb USB







Vzw VoLTE test plans

Device Test Plan VoIP quality

•PESQ testing with VoLTE link

Compliance Test Plan IMS VoIP

•VoLTE call processing, voice function, timers,

InterRAT

Compliance Test Plan LTE RCS

•Video functional, EAB

3

1

2







Mode-sets explained

AMR: 8kHz sampling, 13 bit per sample,

typically 20ms frames

Mode Bit rate kbps Quality

0 4.75 OK

1 5.15

2 5.9

3 6.7

4 7.4 Mode Bit rate kbps Quality

5 7.95 0 6.6 Good

6 10.2 1 8.85

7 12.2 Good 2 12.65 V. Good

3 14.25

4 15.85

5 18.25

6 19.85

7 23.05

8 23.85 Excellent

AMR-WB: 16kHz sampling, 14 bit per

sample, typically 20ms frames

AMR-WB gives better speech quality than

AMR at a given bit rate.

SDP examples

Example 1; use AMR and only allow 12.2kbps


a=fmtp:97 octet-align=1; mode-set=7

Example 2; use AMR and allow several rates


a=fmtp:97 octet-align=1; mode-set=2,3,5,5,6,7

Notes:

•If client A suggests mode-set=7 and client B suggests mode-

set=2,3,5,5,6,7 then both will use only 7.

•If more than one mode is agreed then mode is set on an RTP

frame by RTP frame basis

•Octet-align=1 means octet align

•Octet-align=0 (or not present) means bandwidth efficient RTP

encoding

•Vzw recommends Octet align=0 and only AMR mode 7 or

AMR-WB mode 2

RTP/UDP/IP

AMR AMR

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Business

8 the path to voice over lte - vo lte