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The Path to Voice over LTE - VoLTE
with IMS (IP Multimedia Subsystem)
presented by
Andjela Ilic-Savoia
Agilent Technologies
© 2012 Agilent Technologies
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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 Agilent Technologies
Wireless Communications
3 © 2013 Agilent Technologies
Wireless Communications
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Accelerate next-generation wireless.
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 Agilent Technologies
Wireless Communications
4 © 2013 Agilent Technologies
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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.
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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
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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
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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!
© 2012 Agilent Technologies
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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?
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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)
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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.
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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
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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
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Agenda:
• Presentation:
• 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?
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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
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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,
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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 Agilent Technologies
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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.
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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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GSM/W-CDMA/
TDSCDMA
1xRTT/eVDO/
eHRPD
Phone x Network Configuration
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
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GSM/W-CDMA/
TDSCDMA
1xRTT/eVDO/
eHRPD
Phone x Network Configuration
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
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GSM/W-CDMA/
TDSCDMA
1xRTT/eVDO/
eHRPD
Phone x Network Configuration
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
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GSM/W-CDMA/
TDSCDMA
1xRTT/eVDO/
eHRPD
LTE
Phone x Network Configuration
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
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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?
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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
• …
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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
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continued…QoS class identifier : QCI
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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
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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
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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#
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OTA Messages example:
SPS and TTI Bundling
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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.
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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
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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
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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
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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 Agilent Technologies
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EXTRAs:
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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
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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
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Simple VoLTE Setup
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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
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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
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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
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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
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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
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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=rtpmap:97 AMR/8000
a=fmtp:97 octet-align=1; mode-set=7
Example 2; use AMR and allow several rates
a=rtpmap:97 AMR/8000
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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