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VoLTE Fundamental

Fuzhou, 2015-01-08

Company Confidential
1 © 2015 NSN VoLTE Fundamental Wokshop / Fuzhou / 2015-01-01
VoLTE Introduction
What is VoLTE?

• LTE (Long term Evolution) aka 4G, is the wireless communications


standard developed by 3GPP standardization, to offer high speed with
low latency.
• LTE is packet-switched network only.
• LTE network architecture does not include the traditional 2G/3G voice
services.
• Voice over LTE is the solution to provide the Voice service capability
based on GSMA IR.92 specification.
- Real time traffic
- Quality of Service
- Interoperability to existing CS voice network

• IP Multimedia Subsystem (IMS) enables integrated voice, data and


multi-media services interworking between different access networks. IMS
is based on SIP call control for creating, modifying, and terminating
sessions.
VoLTE Network Architecture Overview

CSCF(Call Session Control Function): Signaling control during multimedia call session
MGCF(Media Gateway Control Function): Perform IMS and CS domain interworking ;
Protocol conversion between different domains
MGW(Media Gateway): User planes connecting different domains;
codec conversion between different networks
VoLTE Bearer Combination

The VoIP service has specific bearer combination requirements:


- QCI1 dedicated bearer for voice call.
- QCI2 dedicated bearer for video call.
- QCI5 dedicated bearer for SIP signalling to the IMS.
- Default non-GBR bearer (QCI8-9) for data transfer (always-on).

QCI Guarantee Priority Delay budget Loss rate Application

1 GBR 2 100 ms 1e-2 VoIP


2 GBR 4 150 ms 1e-3 Video call
3 GBR 5 300 ms 1e-6 Streaming
4 GBR 3 50 ms 1e-3 Real time gaming
5 Non-GBR 1 100 ms 1e-6 IMS signalling
6 Non-GBR 7 100 ms 1e-3 Interactive gaming
7 Non-GBR 6 300 ms 1e-6
TCP protocols:browsing, email, file
8 Non-GBR 8 300 ms 1e-6
download
9 Non-GBR 9 300 ms 1e-6
VoLTE Signaling Overview
SIP Protocol Stack

Can we talk? (my IP/RTP Port is xxx)


SIP Phone SIP Phone
OK (my IP/RTP port is yyy)
Application
(Port#) Chat Chat
codec codec
RTP SIP SIP RTP

Transport UDP SCTP TCP TLS TLS TCP SCTP UDP


Network IP Address IP Address
Link Ethernet Ethernet
Physical Cable Internet Cable

Signaling/Message (Control Plane)


Media (User Plane)

- The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol


for creating, modifying, and terminating sessions with one or more participants developed
by IETF – RFC3261.
- SIP is transferred using QCI5 bearer.
VoLTE Registration

Why register:
- Users use IMPU (IP Multimedia
Public Identity) communication
- Establish the correspondence
between the user's current IP and its
IMPU
- Master the user's current location
information and business capabilities
- Authentication and authentication
of the registration process guarantees
network security
VoLTE Registration

1. UE attaches to the network, the


PGW assigns IP address and identifies
P-CSCF to the terminal.
2. UE sends Registration Request to S-
CSCF (via P-CSCF) in REGISTER
message.
3. S-CSCF performs registration
procedures with HSS and acquires
user authentication information and
sends UE a challenge in 401
Unauthorized message.
4. UE calculates the response and sends
it to S-CSCF in REGISTER message.
5. After the authentication has
succeeded the S-CSCF downloads and
stores service control information
from HSS and S-CSCF notifies the
terminal about completed registration
with 200 OK message.
VoLTE Registration

Common problems at this stage:


- The PDN connection request is not
initiated on the terminal side.
- The network sends a PDN connection
rejection.
VoLTE Call Flow

1. MO UE generates an INVITE request,


which is sent to the IMS. Note that MO
UE includes QoS preconditions in the SDP
which indicates that MO UE does not
want MT UE to be alerted until there are
network resources reserved in both
directions .
2. The P-CSCF acknowledges the INVITE to
the MO UE with "100 Trying” message
indicating that the call setup is in
progress.
3. The SIP method OPTIONS allows a IMS
to query another UE or a proxy server as
to its capabilities. This allows a client to
discover information about the supported
methods, content types, extensions,
codecs, etc. With 200 OK message UE
answers with its capabilities to IMS.
4. The INVITE message received by MT UE
includes declaration for using precondition,
type of media, codec to use and the
protocol for transporting the media in the
SDP.

More detail refer to 3GPP 24228-5f0 7.2.3.1/7.4.3.1


VoLTE Call Flow
5. When precondition mechanism is
supported P-CSCF would send 183
Session Progress to originating UE which
then compares the terminating UE
capabilities with its own and determines
the codec to be used.
6. Originating UE notifies the terminating
UE using PRACK the selected codec.
200 OK is received from terminating
UE. EPS Bearer Activation follows for
both UEs.
7. The UPDATE messages indicates that
precondition on MO UE side is met.
8. Both terminals confirm the setup of
bearer with QoS according to 200 OK
message and terminating UE start
ringing.
9. The 180 Ringing message is initiated
from MT to IMS and then IMS forwards
the ‘Ringing message’ to MO UE.
10. Once both UEs receive 200 OK, they
ACK it and the SIP session is established
- voice communication starts.
Precondition
For the UEs of both parties, the execution process of establishing a PDP context is independent of each other. This
means that there is no guarantee that the negotiated media session will be established until the resource is
successfully reserved. Therefore, the Precondition function is mainly to ensure that the called party should not
ring before confirming that the local and calling party's resource reservation has been successful, so as to

minimize the ringing of the called party but fail to answer the call.

If the Precondition field is carried in the INVITE Request message, the


terminal supports the function.

Call flow that does not support the Precondition function


AMR
• AMR=Adaptive Multi-Rate, "Adaptive Multirate Coding", primarily for audio on mobile devices
• AMR, also known as AMR-NB, has a sampling frequency of 8KHz and a voice bandwidth range of 300-3400Hz.
• AMR-NB has 16 encoding modes, 0-7 corresponds to 8 different encoding modes, and 8-15 is used for noise or
reservation.

• AMR-WB=Adaptive Multi-rate-Wideband, “Adaptive Multi-Rate Wideband Coding” with a sampling frequency of 16


kHz and a voice bandwidth range of 50-7000 Hz
• AMR-WB is a wideband speech coding standard adopted by the international standardization organizations ITU-T
and 3GPP, also known as the G722.2 standard..

AMR-NB AMR-WB
Frame Mode Mode Frame content (AMR mode, comfort noise, Frame
Mode Mode Frame content (AMR-WB mode, comfort
Type Indication Request or other) Type
Indication Request noise, or other)
0 0 0 AMR 4,75 kbit/s Index
1 1 1 AMR 5,15 kbit/s 0 0 0 AMR-WB 6.60 kbit/s
2 2 2 AMR 5,90 kbit/s 1 1 1 AMR-WB 8.85 kbit/s
3 3 3 AMR 6,70 kbit/s (PDC-EFR) 2 2 2 AMR-WB 12.65 kbit/s
4 4 4 AMR 7,40 kbit/s (TDMA-EFR) 3 3 3 AMR-WB 14.25 kbit/s
5 5 5 AMR 7,95 kbit/s 4 4 4 AMR-WB 15.85 kbit/s
6 6 6 AMR 10,2 kbit/s 5 5 5 AMR-WB 18.25 kbit/s
7 7 7 AMR 12,2 kbit/s (GSM-EFR) 6 6 6 AMR-WB 19.85 kbit/s
8 - - AMR SID 7 7 7 AMR-WB 23.05 kbit/s
9 - - GSM-EFR SID 8 8 8 AMR-WB 23.85 kbit/s
10 - - TDMA-EFR SID 9 - - AMR-WB SID (Comfort Noise Frame)
11 - - PDC-EFR SID 10-13 - - For future use
12-14 - - For future use 14 - - speech lost
15 - - No Data (No transmission/No reception) 15 - - No Data (No transmission/No reception)
Media Negotiation

Media consultation:
The calling and called UEs need to agree on the type and encoding mode of the media
during the establishment of the session. For this purpose, the SDP request and response
mechanism is used to negotiate the media; the media types negotiated by the two parties
include video, audio, text, and the like; Media negotiation process:

Supported
speech coding
types
VoLTE Audio Call

Call flow supporting Precondition


VoLTE Video Call
VoLTE Call Flow

Legacy issues at this stage:


- The called VOLTE terminal falls back to GSM, but
it is still not connected (the core network has not
yet been located)
- The terminal strongly locks 4G, causing the
called VOLTE terminal to receive CS Paging or not
receiving a paging message.
VOLTE Performance
MOS value: HD>OTT> SD;
Resource occupation: HD ≈ standard clear
<OTT;
QCI1 < QCI9 ;
End-to-end delay: HD ≈ standard clear
<OTT;
Access delay: VoLTE(2s)<GSM<CSFB;

Medium shift has confirmed the use of high-


definition voice

MOS, PLR, Delay, downlink RB number, downlink MCS are directly related to SINR:
When the SINR is lower than 15, the MOS value is frequently jittered due to inter-cell handover and TM mode
switching. When the SINR is lower than 3, the MOS value begins to decrease significantly, and other indicators begin to
deteriorate to different degrees. ;

高清-MOS/PLR随SINR变化趋势图 高清-Delay随SINR变化趋势图 高清-下行RB数/MCS随SINR变化趋势图


24 4.5 340 12 25
22
4
20 320
10 20
3.5
18
300
16 3 8
14
2.5 280 15
12 6
2 260
10
10
8 1.5
240 4
6
1
4
2 5
0.5
220
2
0 0 200 0 0
14 12 10 8 6 4 2 0 -2 -4 -6 -8 -10 -12 20 18 16 14 12 10 8 6 4 2 0 -2 -4 -6 -9

Packet Loss Rate MOS 端到端时延 下行RB数 下行MCS


eSRVCC Introduction
What is SRVCC?

• SRVCC =Single Radio Voice Call Continuity


• SRVCC functionality is standardized by the 3GPP and the idea behind
this feature is to handover ongoing VoIP call from LTE to another RAT
with simultaneous change of the domain that serves the connection –
from PS to legacy CS
3GPP R8/R9 solution
eSRVCC Network architecture
• The eSRVCC architecture introduces an anchor point on the access side in the visited network. The anchor point
includes two parts of the logical entity:
• - ATCF : Access Transfer Control Functionality
• - ATGW: Access Transfer Gateway
• The ATCF always participates in the session control plane during the call and after the access side transition. When the
eSRVCC occurs, it is responsible for controlling the conversion from the local CS domain new media channel, without
updating the remote call branch, and controlling the ATGW to anchor the media plane. The local media plane is switched to
the MGW, and the remaining media faces remain unchanged. .

3GPP R10 solution


More detail refer to 3GPP 23216 6.2.2.1
eSRVCC Call Flow
P-CSCF ATCF / I-/S-CSCF CSCF UE
MME (A-Party) ATGW (A-Party)
UE (A-Party) eNB
BSC/RNC MSC-S SCC-AS (B-Party) (B-Party)
PDN-GW (A-Party) (A-Party)
MGW

0. MME receives STN-SR


Network Attach and Default EPS Bearer Setup as part of the User Profile. IMS Registration (SIP) IMS Registration (SIP)
IMS Voice Call Setup (SIP)

Dedicated EPS Bearer Setup for VoIP packets


Dedicated EPS Bearer Setup for VoIP packets Media Path (voice)

Measurement the MSC-S reserves resources the ATCF is in the call setup and reserves resources at
PS-Bearer Splitting (not shown)
Reports 2. Handover in the RAN and also at the MGW the AGW (anchors the bearer) at the SIP Session Setup
1. eNB: HO Decision Required 3.PS-to-CS Request (IMSI, Target-ID, STN-SR)

4. Handover Request / Ack ATCF configures ATGW with SDP=MGW


5. SIP Invite (STN-SR, SDP=MGW)
11. UE switches which switches the bearer from
from LTE to 6. SIP 200 OK PDN-GE (LTE) to the MGW (CS)
2G/3G 8. PS-to-CS Response
9. Handover Command 7. SIP ACK Since there is no SDP update, no
10. Handover Command 16. SIP Invite (without SDP update) SIP Re-Invite is sent to the B-Party.
Media Path (voice)

12. Handover Complete 17. SIP 200 OK


the MME initiates the release of
18. SIP ACK after a timeout, the SCC-AS initiates the
the Dedicated EPS Bearer . 13.PS-to-CS Complete / Ack release the source access leg of the session.
PCRF
Delaying the SIP BYE request enables the UE
14.Session Termination Indication to add Gm control (e.g. for video) and to reuse
Deactivation of Dedicated EPS Bearer for VoIP
(with PS to CS handover indicator) the dialog in case of SRVCC cancellation.
19. SIP BYE

15. MAP Update


The UE will receive the SIP BYE only if the Gm-interface is used over Location
the 3G packet bearer after the PS-CS access transfer is completed. in HSS/HLR
eSRVCC Optimization Case

Legacy issues at this stage:


- The eSRVCC user plane interrupt latency is
approximately 600ms, which is much higher than the
medium shift 300ms requirement.
- Redirecting measurement and control cancellation
is not supported during VOLTE calls. The VOLTE voice
drop caused by redirection is easy to occur on the
cell coverage edge, which affects the eSRVCC user
perception success rate.
Suggestions for Optimization in the future
Grid road optimization:
VoLTE voice quality (MOS) is directly related to end-to-end PLR and delay. More attention should be
paid to the following points:
Less inter-cell ping-pong switching
Less TM mode switching
Avoid quality difference sections with SINR below 3dB
KPI indicator:
For the PS drop rate indicator, the inactivity timer in the current network is set to 10s, and the
new formula of the Nokia drop rate masks a large number of wireless problems on the live network. The
rapid rebuild causes the dropped line to not affect the user perception; but for VOLTE voice, it does
not switch. Or the quality difference dropped directly affects customer perception. To avoid a large
number of user complaints, the following points should be noted:
It is not feasible to modify the CIO and other unconventional methods to improve the switching
success rate.
Neighboring area and eSRVCC neighboring area maintenance in LTE system are included in daily work
Wireless planning:
Considering the high failure rate of Fuzhou site, more reasonable neighboring area planning is
necessary.
Considering 5%-10% of different system switching, GSM industrial parameters in the eSRVCC neighboring
area need to be updated in time.
Appendix
Thank You