8 the path to voice over lte - vo lte

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The Path to Voice over LTE - VoLTE
with IMS (IP Multimedia Subsystem)
presented by
Andjela Ilic-Savoia
Agilent Technologies

Wireless Communications
© 2012 Agilent Technologies
2013
1

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?

2013
2

Agenda:

considerations


2013
3

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.

2013
4

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
2013
5

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

2013
6

Enter IMS …

Characteristic

CS Mobile
Telephony

Legacy IP

Standards bodies

3GPP, 3GPP2

IEEE

3GPP/3GPP2 and
IEEE

CS voice

PS data

PS data and PS voice

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

Centralised/ hierarchical/
closed

Distributed/ flat/open

IMS “walled garden”
debate

QoS

Guaranteed bandwidth

Best Effort

GBR and BE
supported

AAA







~5 billion

>500M

Will be lots!

Primary service
Addressing
technique

Architecture

No. of devices

IMS
IP Multi-Media
Service

2013
7

Agenda:

considerations


2013
8

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)

2013
9

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.

2013
10

SIP and IMS

SIP server

IP Network

PSTN
GSM

IP Network

3G

Telephone Network
• 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…)
2013
11

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

2013
12

Agenda:

• Presentation:
considerations

2013
13

IMS uses SIP
(Session Initiation Protocol)
User A

CSCF Proxy

REGISTER

User B
REGISTER

OK

OK


Optional AKAv2 authentication and IPsec

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

SDP (Session Description Protocol)

INVITE

e.g. m (media), a (attribute) etc.

TRYING

INVITE
TRYING

m=audio 49120 RTP/AVP 98 97
a=rtpmap:98 AMR/8000
a=fmtp:98 mode-set=7

RING

RING

OK

OK

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

ACK

ACK

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

RTP/RTCP

BYE
ACK

BYE
ACK

CSCF = Call Session Control Function
2013

14

IMS uses SIP
(Session Initiation Protocol)
User A

CSCF Proxy

REGISTER


User B
REGISTER

CSCF Proxy

OK

OK
INVITE
TRYING

INVITE
TRYING
RING

RING

OK

OK

IP

ACK

ACK

ACK

User A

User B

SIP messages go via proxy server
RTP voice traffic ~ peer to peer (or via
other network nodes)

RTP/RTCP

BYE

Router/IP
network

BYE
ACK

CSCF = Call Session Control Function

2013
15

RTP

RTP is used for the delivery of the user data,

2013
16

Agenda:

considerations

2013
17

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.

2013
18

Phone x Network Configuration

1xRTT/eVDO/
eHRPD

LTE
GSM/W-CDMA/
TDSCDMA

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

e.g.

LTE/eVDO/HRPD radio
GUI
1xRTT CS radio

CS Voice Client
2013

19


1xRTT/eVDO/
eHRPD

LTE
GSM/W-CDMA/
TDSCDMA

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)

e.g.
LTE/W-CDMA/GSM radio
GUI
CS Voice Client
2013
20


1xRTT/eVDO/
eHRPD

LTE
GSM/W-CDMA/
TDSCDMA

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)

E.g.


IMS Voice Client
GUI

1xRTT CS radio

CS Voice Client
2013

21


1xRTT/eVDO/
eHRPD

LTE
GSM/W-CDMA/
TDSCDMA

SRVCC Single Radio Voice Call Continuity
•IMS voice calling in LTE coverage areas
•Quickly handover from LTE to legacy 2G/3G at LTE edge

e.g.

IMS Voice Client
GUI
CS Voice Client
2013

22


1xRTT/eVDO/
eHRPD

LTE
GSM/W-CDMA/
TDSCDMA

SVLTE

CSFB

Simultaneous Voice & LTE
Aka Dual Standby

Circuit Switched
Fall Back

e.g.

LTE/ IMS
islands

SRVCC

LTE / IMS only

Single Radio Voice
Call Continuity

Most Operators will skip some steps
World phones will need to roam with many network configurations
LTE/eVDO/HRPD/GSM/WCDMA/ TDSCDMA radio

1xRTT CS radio

IMS Voice
Client
GUI
CS Voice Client
2013

23

Agenda:

considerations


2013
24

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

• …

2013
25

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

2013
26

continued…QoS class identifier : QCI

2013
27

Protocol Features: Traffic Flow Template

IP packets

Applications

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
Example configuration
Internet
Dedicated

IMS-SIP

RTP

Non-GBR, low error
IPv4/6
address 1

IPv6
address 2

TFT

e.g. IMS-SIP signalling
IP1

IP2
SIP

LTE Radio

GBR, high error

Default
e.g. Browser

e.g. RTP voice stream

RF UE
2013
28

Protocol Features:
GBR, TFT, QoS, DRB’s signalled to UE


Dedicated bearer, linked to
default bearer
GBR

TFT

Negotiated
QoS

2013
29

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.

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

DL after
SPS is
scheduled

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

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

SPS allocation – reduced PDCCH signalling
2013
30

Protocol Features: TTI bundling

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.

NACK

NACK

NACK

ACK

DL: TTI

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 6 7 8 9

UL: RV#

0

DL: TTI

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 6 7 8 9

UL: RV#

0 1 2 3

1

2

3

0 1 2 3
ACK

0 1 2 3

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.

ACK

2013
31

OTA Messages example:
SPS and TTI Bundling


2013
32

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.

2013
33

Protocol Features: RoHC continued
Cuts IP overhead e.g. RTP streams for speech;
2:1 for IPv4, 3:1 for IPv6

•

RoHCv2 simplification robustness handling of outof-sequence packets


Applications

•

Payload

DeCompressor
RoHC
Context

Payload

Header

Payload

Compressor
RoHC
Context

LTE Radio Layers

Payload

LTE Radio Layers

LTE Radio

Header

IP packets

Network

RF UE
2013
34

Protocol Features:
SigComp - SIP Signalling Compression

SIP/SDP signalling
stack
State full Compressor
/De-compressor

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

2013
35

Protocol Features: IP Sec

Tunnel Mode (VPN like)
• Entire original IP datagram encrypted

More info in
RFC4301, 4302, 4303

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

Normal IP traffic

IP Header
IP Header

Transport Mode
Tunnel Mode

New IP Hdr

AH

AH

Data
Data

IP Header Data

2013
36

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

2013
37

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

2013
38

EXTRAs:

2013
39

CS vs. PS

LTE PS

IMS core
emulation
RF

IP

Audio

2G/3G PS

RF

IP

RF

IMS end-point
emulation

Audio

2G/3G CS

2013
40

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

2013
41

Simple VoLTE Setup

2013
42

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

Delay/Jitter/Loss insertion

Agilent IMS-SIP server
Ethernet

Agilent PXT

Agilent PXT
VoLTE UE

VoLTE UE
RF

RF
Audio Analyzer

Audio in/out headphone jack

Audio in/out headphone jack
2013

43

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

Agilent IMS server and
several remote clients

Agilent 8960
1xRTT cell

Ethernet

Agilent 8960
eHRPD cell

Test
automation

Agilent PXT
LTE cell

RF

2013
44

Delay and Jitter insertion with ZTI NetDisturb

ZTI NetDisturb

Agilent IMS-SIP server & client(s)
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

Windows XP or 7 PC

Ethernet

VoLTE UE
Delay/Jitter
insertion

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

Agilent PXT

RF
2013

45

Agilent-B&K VoLTE Audio Test System

Agilent IMS-SIP server & client

.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

B&K ZE 0948

Windows 7 PC

USB

VoLTE UE
Ethernet
Audio Line
in/out

Agilent PXT

RF

HATS

B&K PESQ Audio Analyzer

2013
46

Adding Jitter to audio quality systems
Agilent IMS-SIP server & client

.bat file, IPv6
router adv*

Delay/Jitter tool,
e.g. ZTI NetDisturb

USB

B&K Audio Test Software
Optional
Delay/Jitter added
to any link
B&K ZE 0948

Windows 7 PC

USB

VoLTE UE
Ethernet
Audio Line
in/out

Agilent PXT

RF

HATS

B&K PESQ Audio Analyzer

2013
47

Vzw VoLTE test plans

Device Test Plan VoIP quality

1

•PESQ testing with VoLTE link

Compliance Test Plan IMS VoIP

2

•VoLTE call processing, voice function, timers,
InterRAT

Compliance Test Plan LTE RCS

3

•Video functional, EAB

2013
48

Mode-sets explained

SDP examples
AMR

AMR

RTP/UDP/IP

Example 1; use AMR and only allow 12.2kbps
a=fmtp:97 octet-align=1; mode-set=7

AMR: 8kHz sampling, 13 bit per sample,
typically 20ms frames
Mode
0
1
2
3
4
5
6
7

Bit rate kbps
4.75
5.15
5.9
6.7
7.4
7.95
10.2
12.2

Example 2; use AMR and allow several rates
a=fmtp:97 octet-align=1; mode-set=2,3,5,5,6,7

Quality
OK

AMR-WB: 16kHz sampling, 14 bit per
sample, typically 20ms frames

Good

Mode
0
1
2
3
4
5
6
7
8

Bit rate kbps Quality
6.6
Good
8.85
12.65
V. Good
14.25
15.85
18.25
19.85
23.05
23.85
Excellent

AMR-WB gives better speech quality than
AMR at a given bit rate.

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

2013
49

8 the path to voice over lte - vo lte

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