8 Thepathtovoiceoverlte Volte 131205064856 Phpapp02 PDF

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 Greater insight. Greater confidence.
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 Greater insight. Greater confidence.
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 Legacy IP IMS Accelerate next-generation wireless.
Telephony 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 Phone numbers IP addresses IP + support for legacy
technique 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/ Distributed/ flat/open IMS walled garden
closed debate
QoS Guaranteed bandwidth Best Effort GBR and BE
supported
AAA
No. of devices ~5 billion >500M Will be lots!
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 Greater insight. Greater confidence.
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 Accelerate next-generation wireless.
PSTN
IP Network
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) Greater insight. Greater confidence.
User A CSCF Proxy User B
Optional AKAv2 authentication and IPsec
REGISTER
REGISTER
OK SIP (Session Initiation Protocol)
OK e.g. INVITE, TRYING, RING, OK, BYE etc.
INVITE SDP (Session Description Protocol)

e.g. m (media), a (attribute) etc.
TRYING
INVITE m=audio 49120 RTP/AVP 98 97
TRYING 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
BYE
ACK
ACK CSCF = Call Session Control Function
2012
14
IMS uses SIP
(Session Initiation Protocol)
User A CSCF Proxy User B
REGISTER CSCF Proxy
REGISTER
OK OK
INVITE
TRYING
INVITE
TRYING
RING Router/IP
RING network
OK
OK
IP
ACK User A User B
ACK
SIP messages go via proxy server

RTP/RTCP RTP voice traffic ~ peer to peer (or via
other network nodes)
BYE
BYE
ACK CSCF = Call Session Control Function
ACK
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.
LTE/W-CDMA/GSM radio IMS Voice Client
GUI
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.
LTE/W-CDMA/GSM radio IMS Voice Client
GUI
CS Voice Client
2012
22
1xRTT/eVDO/
eHRPD
LTE
GSM/W-CDMA/
TDSCDMA
SVLTE CSFB LTE/ IMS SRVCC LTE / IMS only

Simultaneous Voice & LTE Circuit Switched Single Radio Voice
islands
Aka Dual Standby Fall Back Call Continuity
Most Operators will skip some steps

World phones will need to roam with many network configurations
e.g.
LTE/eVDO/HRPD/GSM/W- IMS Voice
CDMA/ TDSCDMA radio
Client GUI
2012
23
Agenda:
considerations
2012
24
LTE Protocol features that will affect
how VoLTE performs Greater insight. Greater confidence.
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 Greater insight. Greater confidence.
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
o TFT is set of packet filters ~ EPS bearer

o TFT decides on IP traffic priority
Applications
o Sort IP packets based on:
IP protocol; e.g. UDP, TCP
Port number
IP address
Priority
IP packets
Example configuration
Internet IMS-SIP e.g. RTP voice stream
Dedicated GBR, high error

RTP
Default
LTE Radio
Non-GBR, low error
IPv4/6 IPv6
TFT
e.g. Browser e.g. IMS-SIP signalling
address 1 address 2
IP1
IP2
SIP
RF UE
2012
28
Protocol Features:
GBR, TFT, QoS, DRBs signalled to UE Greater insight. Greater confidence.
Dedicated bearer, linked to

default bearer
GBR
TFT
Negotiated
QoS
2012
29
Protocol Features:
Semi-persistent scheduling (SPS) Greater insight. Greater confidence.
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 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
DL after
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 is
scheduled
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 Normal transmission with HARQ feedback

delay of 4ms for each 8ms HARQ cycle up
UL: RV# 0 1 2 3
to 28ms for final 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 Bundled TTI HARQ process 0 in 4

consecutive TTIs (different RV). Total
UL: RV# 0 1 2 3 0 1 2 3 0 1 2 3
feedback delay is only 8ms.
ACK ACK
2012
31
OTA Messages example:
SPS and TTI Bundling Greater insight. Greater confidence.
2012
32
Protocol Features:
ROHC - Robust Header Compression Greater insight. Greater confidence.
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; Greater insight. Greater confidence.
Applications
2:1 for IPv4, 3:1 for IPv6
RoHCv2 simplification robustness handling of out-

of-sequence packets
Network
IP packets
Header Payload Header Payload
De- Compressor
Compressor
RoHC RoHC
Context Payload Context Payload
LTE Radio
LTE Radio Layers LTE Radio Layers
RF UE
2012
34
Protocol Features:
SigComp - SIP Signalling Compression Greater insight. Greater confidence.
Compresses text based SIP and SDP

messages
SIP/SDP signalling RTP/RTCP
stack Media stack Up to 3:1 compression
Standardized by the IETF RoHC working
State full Compressor group
/De-compressor
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 Data
Transport Mode IP Header AH Data
Tunnel Mode New IP Hdr AH 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
IMS core
emulation
LTE PS RF IP
Audio
IMS end-point
2G/3G PS RF IP emulation
RF Audio
2G/3G CS
2012
40
Agilent E6966A 1FP & 2FP IMS-SIP client
Easy to install and use Windows 7 PC-based IMS-SIP client Accelerate next-generation wireless.
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
Test
automation
Agilent IMS server and Ethernet
several remote clients
Agilent 8960 Agilent 8960 Agilent PXT

1xRTT cell eHRPD cell LTE cell
RF
2012
44
Delay and Jitter insertion with ZTI NetDisturb
ZTI NetDisturb
Agilent IMS-SIP server & client(s)
USB-Ethernet adaptors can be used with ZTI
NetDisturb to add delay/Jitter to any Ethernet link in
USB 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 Greater insight. Greater confidence.
* Batch file can be used to configure Windows 7 PC

.bat file, IPv6 to transmit IPv6 Router Advertisements. Alternatively
an IPv6 router can be added to the connection
router adv* between a Windows 7 or XP PC and the PXT.
B&K Audio Test Software
Windows 7 PC
B&K ZE 0948 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*
USB
Delay/Jitter tool,
e.g. ZTI NetDisturb
B&K Audio Test Software
Optional
Delay/Jitter added
to any link
USB
Windows 7 PC
B&K ZE 0948
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 1
Compliance Test Plan IMS VoIP
VoLTE call processing, voice function, timers, 2
InterRAT
Compliance Test Plan LTE RCS 3

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=rtpmap:97 AMR/8000
a=fmtp:97 octet-align=1; mode-set=7
AMR: 8kHz sampling, 13 bit per sample,
typically 20ms frames Example 2; use AMR and allow several rates
Mode Bit rate kbps Quality a=rtpmap:97 AMR/8000
0 4.75 OK a=fmtp:97 octet-align=1; mode-set=2,3,5,5,6,7
1 5.15
2 5.9 AMR-WB: 16kHz sampling, 14 bit per
3 6.7 sample, typically 20ms frames
4 7.4 Mode Bit rate kbps Quality
5 7.95 0 6.6 Good
6 10.2 1 8.85 Notes:
7 12.2 Good 2 12.65 V. Good If client A suggests mode-set=7 and client B suggests mode-
3 14.25 set=2,3,5,5,6,7 then both will use only 7.
4 15.85 If more than one mode is agreed then mode is set on an RTP
5 18.25 frame by RTP frame basis
Octet-align=1 means octet align
6 19.85
Octet-align=0 (or not present) means bandwidth efficient RTP
7 23.05
encoding
8 23.85 Excellent Vzw recommends Octet align=0 and only AMR mode 7 or
AMR-WB mode 2
AMR-WB gives better speech quality than
AMR at a given bit rate.
2012
49

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Greater insight. Greater confidence.

Accelerate next-generation wireless.

The Path to Voice over LTE - VoLTE

3GPP defined IMS in 1999

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

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

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

CSFB (Circuit Switched Fall Back) - single radio approach

Architecture divided into:

CSCF is further divided into nodes:

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

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

IMS/VoLTE capable device

INVITE SDP (Session Description Protocol)

SIP messages go via proxy server

RTP is used for the delivery of the user data,

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

SVLTE Simultaneous Voice & LTE aka Dual Standby

CSFB Circuit Switched Fall Back

LTE / IMS Islands

SRVCC Single Radio Voice Call Continuity

SVLTE CSFB LTE/ IMS SRVCC LTE / IMS only

Most Operators will skip some steps

Strict packet delay-based QoS

o TFT is set of packet filters ~ EPS bearer

o TFT decides on IP traffic priority

Dedicated GBR, high error

Dedicated bearer, linked to

For highly repeatable applications such

In voice we normally only need a few

SPS is used to tell the UE to use specific

SPS allocation reduced PDCCH signalling

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 Normal transmission with HARQ feedback

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 Bundled TTI HARQ process 0 in 4

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

RoHCv2 simplification robustness handling of out-

Compresses text based SIP and SDP

Normal IP traffic IP Header Data

Transport Mode IP Header AH Data

Tunnel Mode New IP Hdr AH IP Header Data

GSMA IR.92 IMS Profile for Voice and SMS

The Path to Voice over LTE - VoLTE

IPv4, IPv6, Voice, video, SMS, GUI

Agilent IMS-SIP server

Agilent PXT Agilent PXT

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

Agilent 8960 Agilent 8960 Agilent PXT

* Batch file can be used to configure Windows 7 PC

B&K Audio Test Software

B&K PESQ Audio Analyzer

.bat file, IPv6

B&K Audio Test Software

B&K PESQ Audio Analyzer

Device Test Plan VoIP quality

Compliance Test Plan LTE RCS 3

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