Media Client - JsSip
WebRTC Device:
Device Status
Registration
Connect to Server
Register
Privacy Permission
Mic Access
Camera Access
Voice Call
Voice Call Controls
A
B
00:00
Mute
Keypad
Hold
Call
Keypad
1
2
a b c
3
d e f
4
g h i
5
j k l
6
m n o
7
p q r s
8
t u v
9
w x y z
*
0
+
#
Video Call
Video Call Controls
Start Video Call
Mute Camera
Video Display
SIP INFO Messages
SIP INFO Messages
Content-Type
text/plain
application/json
application/x-www-form-urlencoded
Message Body
Genesys User Data
Send
Message History
Workspace Tab Bar
Audio & Video Statistics
General
Local Ip -
Remote Ip -
Audio
Video
Codec
-
-
Network
-
-
RTT (ms)
-
-
Port
-
-
Packets Received (/sec)
-
-
Packets Sent (/sec)
-
-
Send
Receive
Bandwidth (kb/s)
Audio
-
-
Video
-
-
Audio
Jitter (ms)
-
-
Packet loss
-
-
Video
Jitter (ms)
-
-
Packet loss
-
-
Resolution
-
-
Frames /sec
-
-
Account Config
Preferences
Config Source
Config Source
Load from Web API
Load From File
Choose New File to Load
Export
Refresh
Call WS
Basic Settings
Account Basic Settings
Device Username:
Password:
Realm / Domain:
Display Name:
Auth Token:
Web Socket Server:
Extended Settings
Account Extended Settings
Business Unit:
App Context:
Client Session Unique Identifier:
Caller Asserted Identity (Caller ID):
Default Destination User (To):
Web Service Settings
Web Service Settings
Ephemeral Device Name:
Business Unit:
App Context:
Client Session Unique Identifier:
Call Behavior
Call Type
Audio
Video
Transceiver Direction
send and receive
send only
receive only
inactive
Video Quality
HD (720x720px)
HD (1280x720px)
VGA (480x480px)
VGA (640x480px)
QVGA (240x240px)
QVGA (320x240px)
Call Behavior
Auto Answer
Auto Answer After (Seconds)
Initiation / Answer with Video
DTMF Behavior
Send Via DTMF 2833
Load From File
DTMF 2833 and SIP INFO
Playback DTMF Tone
3rd Party Call Control
Answer / Resume (Talk)
Hold / Resume (Hold)
SIP NOTIFY is supported for 3rd Party Talk and Hold. NOTIFY (event talk) will answer or retrive the call. NOTIFY (event hold) will the call on hold.
Device Controls
Device Selection
Speaker:
Microphone:
Camera:
Volume Controls
Master Volume
Ringer Volume
Tones Volume
Preview Volume
Call Volume
Codecs
Audio Codecs
Opus
PCMU
PCMA
Video Codecs
H264
VP8
Headers
Custom Headers
+
Troubleshooting
Call Statistics
Audio & video statistics
Log Settings
Log Level:
Debug
Release
Start Logging
Show SIP Message Dialog Pane
Show Call Health / Status Pane
~/webrtc-capture/webrtc_10:10:30.log
Clear Log
Suppress Log Events
Select Events to Suppress from Log
JsSIP Emitters
Media Client Emitters
SIP Messages
Call Stats
Stream Capture
Capture Incoming Audio
~/webrtc-capture/audio_out_1234.wav
Capture Outgoing Audio
~/webrtc-capture/audio_out_1234.wav
Capture Incoming Video
~/webrtc-capture/audio_out_1234.wav
Capture Outgoing Video
~/webrtc-capture/audio_out_1234.wav
Audio & Video Statistics
-
-
Audio
Video
Codec
-
-
Network
-
-
RTT (ms)
-
-
Port
-
-
Packets Received (/sec)
0
0
Packets Sent (/sec)
0
0
Send
Receive
Bandwidth (kb/s)
Audio
0
0
Video
0
0
Audio
Jitter (ms)
0
0
Packet loss
0
0
Video
Jitter (ms)
0
0
Packet loss
0
0
Resolution
0
0
Frames /sec
0
0
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