Webex Contact Center with WebRTC
Task 1 - Configuring WebRTC on Webex Contact Center
Please use your tenant to complete this lab.
Info
Webex Contact Center is introducing Web Real-Time Communication (WebRTC) support for Agent Desktop using the Next Generation Media Platform (RTMS). With this feature, agents can use Agent Desktop with a headset without an external phone or extension number. Agent Desktop supports all current voice functionalities such as hold, retrieve, transfer, and conference. Features such as mute, auto-answer, and dial pad are added to Agent Desktop to facilitate browser-only use.
Configuring WebRTC capability for Webex Contact Center agent
- Navigate to Control Hub and log in with the credentials provided above.
- After logging in to Control Hub, navigate to the 'Contact Center' menu under 'Services'.
- Navigate to 'Voice' and make sure that the 'WebRTC' toggle is enabled.
- Navigate to 'Desktop Profiles' and create a new desktop profile by clicking on 'Create Desktop Profile'.
- Provide a name for the desktop profile.
- Move to 'Outdial options' by clicking 'Next'.
- Enable 'Outdial':
- Select preconfigured 'Outdial Entry Point-1' as an 'Outdial Entry Point'.
- Select preconfigured 'Default_Outdial_ANI' as an 'Outdial ANI'.
- Enable 'Dial Plan'.
- Select 'Any Format' under 'Select Dial Plan'.
- Move to 'Voice Channel Options' by clicking 'Next':
- Make sure that 'Desktop' is enabled under 'Voice Channels options'.
- Move to the end of desktop profile creation by clicking 'Next' and finally 'Create'.
- Navigate and edit your user under the 'Contact Center Users' part.
- Assign the '2008_Team' team under 'Teams'.
- Assign your newly created desktop profile under 'Desktop Profile'.
- 'Save' changes.
- Congratulations, you have completed this task!
Task 2 - Logging in to Agent Desktop Using WebRTC
Step 1 - Login into Agent Desktop
Info
This task is aimed to demonstrate initial WebRTC issues that could be faced by agents when trying to register Agent Desktop using the WebRTC option.
- Open Mozilla Firefox or Microsoft Edge web browser.
- Navigate to Agent Desktop and log in with the credentials provided above.
- After logging into Agent Desktop, select the '2008_Team' team and the 'Desktop' option.
- Click 'Submit'.
- An error about an unsupported browser appears.
Note
Currently, the only supported browser for Agent Desktop WebRTC is Google Chrome. Additional browser support is coming in the future.
- Close Mozilla Firefox/Microsoft Edge web browser.
- Open Google Chrome web browser.
- Navigate to Agent Desktop and log in with the credentials provided above.
- After logging in to Agent Desktop, select the '2008_Team' team and the 'Desktop' option.
- Click 'Submit'.
- An error about microphone access appears.
Step 2 - Enabling browser permissions to access the microphone
- Click on the 'View site information' button located on the left side of Chrome's address bar.
- Click on 'Site settings'.
- Under microphone select 'Allow' and close Chrome's 'Setting' tab.
- Reload the Agent Desktop website.
- Make sure the '2008_Team' team and 'Desktop' option is selected.
- Click 'Submit' - log in using the WebRTC option should succeed now.
- Congratulations, you have completed this task!
Task 3 - Exploring WebRTC data and statistics
Info
This task showcases available WebRTC call data and statistics using built-in browser tools and simulates missing audio during the call.
Section 1 - Exploring WebRTC internals
- Set your state to 'Available' on the Agent Desktop.
- Make an inbound call by calling +19842590500 number and answer via Agent Desktop when the call is presented.
- Open a new tab in Google Chrome web browser.
- Navigate to chrome://webrtc-internals.
- In the WebRTC-internals tab select the active WebRTC session part (desktop.wxcc-us1.cisco.com in its name).
- The following information can be seen:
- On the left side (1), you'll find a record of the PeerConnection API activities. The RTCPeerConnection is the central interface in the WebRTC API. It represents the connection between the local and remote peer and provides all the functions and events necessary to establish the connection (more details can be found at https://webrtc.org/getting-started/overview). These traces (1) display interactions with the RTCPeerConnection object and parameters used.
- On the right side (2), you'll find the metrics collected via the getStats API about the current WebRTC connection.
- On the bottom (3), you'll find graphs produced by utilizing the getStats API (2) about the current WebRTC connection.
- Local and remote IPs and ports in use for this particular WebRTC connection can be seen under traces of the PeerConnection API calls.
- Various live call statistics (i.e. sent packets, received packets, latency, dropped packets etc.) related graphs can be seen at the bottom.
- WebRTC log dump can be retrieved by clicking 'Create a WebRTC-Internals dump' and clicking 'Download the "webrtc-internals dump" option.
Note
WebRTC log dump file can be parsed and analyzed using specific tools, i.e. https://fippo.github.io/webrtc-dump-importer
- WebRTC audio can also be captured for diagnostic purposes.
- End the call.
- Change your state to the idle 'Meeting' state on the Agent Desktop.
Section 2 - Investigating missing audio
- Download task3s2.zip file from https://github.com/asuchank/clus24/raw/main/task3s2.zip and extract it.
- Make a new call to +19843690030 using WebRTC via Agent Desktop (music should be playing).
- Navigate to chrome://webrtc-internals in a new tab.
- In the WebRTC-internals tab select the active WebRTC session part (it will have desktop.wxcc-us1.cisco.com in its name).
- At the bottom of the WebRTC-internals diagnostic page expand bolded text to uncover graphs for an ongoing session.
- Right-click on the task3s2.bat file and choose 'Run as administrator' (audio will be lost for 10 seconds).
- Observe packetsReceived/s go down to zero on the expanded graph from before.
- Based on the observation above - missing audio was caused by the lack of media packets.
- End the call.
- Congratulations, you have completed this task!