NUGU call is a hands-free calling platform that allows connections anywhere through NUGU Touch Points. It supports multi-device connections under one account and has voice UX features like initiating and ending calls through voice commands. Voice quality is maintained through standards for loudness, frequency response, and other factors. Real-time communication uses internet protocols and signaling for call setup, media exchange, and termination. Upcoming features will expand NUGU call to support video calls and intelligent contextual commands.
12. NUGU call Voice Quality Standard
Fixed Distance
Objective Standard
• Loudness
• Frequency Response
• Total Harmonic Distortion
• Etc.
Subjective Standard
• Tone
• Naturality
• Intelligibility
• Ambience
• Etc.
16. Basic Call Signaling
Invite (SDP Offer)
Invite (SDP Offer)
100 Trying
180 Ringing
180 Ringing
200 OK (SDP Answer)
200 OK (SDP Answer)
Ack
Ack
RTP
Bye
Bye
200 OK
200 OK
UAC
Alice
UAC
Bob
UAS
(콜서버)
RTP
전화 발신
링톤 재생
통화 수락
보이스 교환 시작
통화 종료
링백톤 재생
* Terminology
UAS : User Agent Server
UAC: User Agent Client
Internet
Signaling
Media
Redirect
Registrar
Application
Signal Stack
Signal Proxy
Load BalancerUser DB
Location
Media Proxy
TURN
STUN
Signaling
Registrar
Application
Signal Stack
Signal Proxy
User DB
Location
UAC UAC
UAS UASSignal
Media Media
Signal
17. Basic Call Signaling
INVITE sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds
Max-Forwards: 70
To: Bob <sip:bob@biloxi.com>
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@pc33.atlanta.com
CSeq: 314159 INVITE
Contact: <sip:alice@pc33.atlanta.com>
Content-Type: application/sdp
Content-Length: 142
Invite from Alice to UAS
SIP/2.0 200 OK
Via: SIP/2.0/TCP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1 ;received=192.168.10.1
Via: SIP/2.0/TCP pc33.atlanta.com;branch=z9hG4bKnashds8 ;received=10.1.3.33
To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@pc33.atlanta.com
CSeq: 314159 INVITE
Contact: <sip:bob@192.168.10.20>
Content-Type: application/sdp
Content-Length: 131
200 OK from Bob to UAS
SDP Offer (in Invite) SDP Answer (in 200 OK)
v=0
o=TEST-IMS-UE 1513311076564347 0 IN IP6 2001:0:0:1::11
s=SS VOIP
c=IN IP6 2001:0:0:1::11
t=0 0
m=audio 1268 RTP/AVP 127 114 113 102 115 105 101
b=AS:38
b=RS:0
b=RR:0
a=rtpmap:127 EVS/16000
a=fmtp:127 br=5.9-13.2;bw=nb-wb;ch-aw-recv=2
a=rtpmap:114 AMR-WB/16000/1
a=fmtp:114 mode-change-capability=2;max-red=220
a=rtpmap:113 AMR-WB/16000/1
a=fmtp:113 octet-align=1;mode-change-capability=2;max-red=2
20
a=rtpmap:102 AMR/8000/1
a=fmtp:102 mode-change-capability=2;max-red=220
a=rtpmap:115 AMR/8000/1
a=fmtp:115 octet-align=1;mode-change-capability=2;max-red=2
20
a=rtpmap:105 telephone-event/16000
a=fmtp:105 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=sendrecv
a=ptime:20
a=maxptime:240
v=0
o=sharetechnote 662 837663 IN IP6 2001:0:0:1::2
s=-
i=A VOIP Session
c=IN IP6 2001:0:0:1::2
t=0 0
m=audio 60000 RTP/AVP 127 105
b=AS:38
b=RS:0
b=RR:0
a=rtpmap:127 EVS/16000
a=fmtp:127 br=5.9-13.2;bw=nb-wb;ch-aw-recv=2
a=rtpmap:105 telephone-event/16000
a=fmtp:105 0-15
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=ptime:20
a=maxptime:240
a=sendrecv
a=rtcp:60001
18. Basic call Voice Flow
Downlink
Uplink
Jitter Buffer Management
Speech Enhancement
Vocoder
(Encoder)
Media
Protocol Encryption Socket
AMR
AMR-WB
G.711
OPUS
RTP
RTCP
SRTP UDP
Internet
Speech Enhancement
Media
Protocol Encryption Socket
RTP
RTCP
SRTP UDP
AEC NS AGC EQ VOL
VOL EQ AGC NS
Vocoder
(Decoder)
AMR
AMR-WB
G.711
OPUS
PCM
JB
PKT
JB
N/W Monitor
20. DTX with VAD / CNG
Fig.1 DTX operation
• DTX : Discontinuous Transmission (= SCR)
• VAD : Voice Activity Detection
• CNG : Comfort Noise Generator
21. Speech Enhancement for Full-Duplex Voice Call
• AEC - Remove the echo from the mixed mic input referring to the play-backed
output
• NS - Noise suppression on voice and non-voice period
• AGC(DRC) - Adjust the voice volume to the constant level
• EQ - Edit the each frequency-band energies
• VOL - Volume controller for the final stage of each SE signal chains
22. AEC (Acoustic Echo Canceller)
Fig.1 Echo Propagation
Fig.2 Simple AEC Block Diagram
Fig.3 Captured Signal from Micro-phone
Fig.4 Echo Removal from the Captured Signal
FE/NE Time Critical Operation
24. AGC (Automatic Gain Controller)
Fig.1 AGC Input/Output Fig.2 AGC Usage
CF) Direct Cut-Off without AGC
25. Role of the Jitter Buffer
PS Network
Hi Where are you?
Hi Whe re are you?
Hi Whe re are you?
a
b
a : Time-Varying Jitter and Delay
b : Constant Play-Back Delay
TX
RX
Play-Back
(No Buffering)
Jitter
PS Network
Hi Where are you?
Hi Whe re are you?
Hi Where are you?
Voice Packet 20ms
a
a : Time-Varying Jitter and Delay
b : Constant Play-Back Delay
c : b + Jitter Buffer Delay (40ms)
TX
RX
Play-Back
(With Jitter Buffer) b 40ms
c
Jitter
No Buffering Fixed Jitter Buffering
26. Adaptive Jitter Buffer Management
In
Out
Delay
Hi Whe re are you?
Adjusting Buffer
JBM
Target
Buffer
Level
Stationary?
Non-Stationary?
Jitter-Spike?
Weak-Network?
Monitoring Pattern
Packet Loss Concealment
or
Time-Stretching
Internet