8 Thepathtovoiceoverlte Volte 131205064856 Phpapp02

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
Wireless Communications
2012
2013 Agilent Technologies
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?
2012
2
Agenda:
considerations

2012
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.
2012
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
2012
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
2012
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
2012
7
Agenda:
considerations

2012
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)
2012
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.
2012
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)
2012
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
2012
12
Agenda:
Presentation:
considerations
2012
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

2012
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
2012
15
RTP
RTP is used for the delivery of the user data,
2012
16
Agenda:
considerations
2012
17
What ifLTE/IMS out of coverage?

Then:
the Voice call to work would require some level or I-RAT.
Lets examine each of the voice related I-RAT behaviors separately.
2012
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
2012
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
2012
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
2012
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
2012
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
2012
23
Agenda:
considerations

2012
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
2012
25
Protocol Features:
VoIP QoS and Multiple PDNs

Strict packet delay-based QoS

QoS will vary by allocation, by
application and will be heavily
dependent on system capacity.
UEs can have multiple data streams,

Multiple PDNs, 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
2012
26
continuedQoS class identifier : QCI

2012
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
2012
28
Protocol Features:
GBR, TFT, QoS, DRBs signalled to UE

Dedicated bearer, linked to

default bearer
GBR
TFT
Negotiated
QoS
2012
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

100s 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

2012
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 TTIs.
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#
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
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 TTIs (different RV). Total
feedback delay is only 8ms.
ACK
2012
31
OTA Messages example:

SPS and TTI Bundling

2012
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.
2012
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
2012
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
2012
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
2012
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
2012
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
2012
38
EXTRAs:
2012
39
CS vs. PS
LTE PS
IMS core
emulation
RF
IP
Audio
2G/3G PS
RF
IP
RF
Audio
IMS end-point
emulation
2G/3G CS
2012
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
2012
41
Simple VoLTE Setup

2012
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

2012
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
2012
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
2012
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
2012
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
2012
47
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
2012
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
2012
49

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Greater insight. Greater confidence.

Accelerate next-generation wireless.

The Path to Voice over LTE - VoLTE

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

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

IMS - IP Multimedia Subsystem

Greater insight. Greater confidence.

3GPP defined IMS in 1999

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

Support for Voice with LTE

2G/3G Circuit Switched calls have an allocated resource even

LTE UE will generally only be provided resources when it is

Allows efficient use of network resources. If we are saying nothing

Voice with LTE: what is the goal and

Greater insight. Greater confidence.

To deliver same standard of voice call with VoLTE as is delivered by

PS data and PS voice

IP + support for legacy

IMS walled garden

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

IMS has quite complex architecture

Architecture divided into:

Greater insight. Greater confidence.

It is assigned to an IMS-capable UE terminal before registration (through OTA

SIP and IMS

To test VoLTE today, you need:

IMS/VoLTE capable device

Somebody to talk to: Either

IMS uses SIP

Greater insight. Greater confidence.

Optional AKAv2 authentication and IPsec

SIP (Session Initiation Protocol)

SDP (Session Description Protocol)

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

m=audio 49120 RTP/AVP 98 97

RTP (Real-time Transport Protocol)

RTCP (RT Control Protocol)

CSCF = Call Session Control Function

IMS uses SIP

Greater insight. Greater confidence.

SIP messages go via proxy server

CSCF = Call Session Control Function

RTP is used for the delivery of the user data,

What ifLTE/IMS out of coverage?

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

Lets examine each of the voice related I-RAT behaviors separately.

Phone x Network Configuration

SVLTE Simultaneous Voice & LTE aka Dual Standby

Phone x Network Configuration

CSFB Circuit Switched Fall Back

Phone x Network Configuration

LTE / IMS Islands

IMS Voice Client

Phone x Network Configuration

SRVCC Single Radio Voice Call Continuity

IMS Voice Client

Phone x Network Configuration

Simultaneous Voice & LTE

LTE / IMS only

Single Radio Voice

Most Operators will skip some steps

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

LTE Protocol features that will affect

Greater insight. Greater confidence.