This article outlines the technical, environmental, and infrastructural factors that influence the handling and quality of ExpertConnect VoiceHub calls between customers (via PSTN) and advisor (via web dashboard and mobile apps - iOS & Android). It provides actionable recommendations for IT groups to ensure a reliable and high-quality calling experience for users.

Overview of Voice Calling

ExpertConnect's VoiceHub service enables voice calls between customers (via PSTN) and dealership advisors (via the web or mobile app). Understanding how calls are routed and handled is key to ensuring a reliable experience.

Call Types

• Inbound Calls: A customer initiates a call to a dealership's VoiceHub number, or a number that forwards into ExpertConnect. This triggers the configured VoiceHub workflow, such as Ring Schedule, Call Tree, or Voicemail, which determines how the call is routed at the team level. Based on team settings, the system rings available advisors. When an advisor answers, the call is connected, and a new ticket is created or an existing ticket is updated with the call record. 
• Outbound Calls: An advisor initiates a call while creating a new ticket or from within an existing ticket using the web dashboard or mobile app. If the customer answers, the call between the two participants is established. 

Call Mediums

• Customer: This call, regardless of what underlying call medium they are using, will always be received as a PSTN call by our partner telephone provider. 
• Advisor: Uses WebRTC (if call was initiated or received on web dashboard) or VoIP (if call was initiated or received on the mobile app - iOS or Android). 

Call Lifecycle

1. Call Streaming: This phase begins once both participants are connected. It focuses on the real-time transmission of audio and encompasses all aspects related to maintaining call quality throughout the conversation.
2. Call Handling: This phase encompasses all programmable logic involved in managing the call, including the execution of VoiceHub workflows and any post-call processes within ExpertConnect. It also includes the seamless integration and hand-off between the ExpertConnect service and partner telephony provider's cloud infrastructure.

Potential Voice Issues

Below are potential issues that users may experience during call streaming. 

Factors that Impact Call Quality

Network Conditions

Bandwidth - Network Conditions 

Our partner telephony provider recommends a minimum of 100 kbps of bandwidth per concurrent voice call (both upstream and downstream) to maintain high-quality audio. This applies to the call leg on the advisor side, which rely on WebRTC and Secure Media (ICE/STUN/SRTP) for media transmission. 

Network Conditions 

Our partner telephony provider recommends keeping one-way latency below 150 milliseconds for most real-time voice communication scenarios. While communication can still function with latency up to 250–300 ms, it may begin to feel unnatural or cause interruptions in conversation flow. 

Breakdown:

• Optimal: Under 100 ms (ideal for interaction sessions)
• Acceptable: 100-150 ms (suitable for most use cases)
• Degraded experience: 250-300 ms (noticeable delays)
• Maximum threshold: 400 ms (ITU-T G.144 guideline for network planning) 

Jitter

There is no specific string numerical threshold identified for jitter in all scenarios, but based on the overall experience and testing, here are the key takeaways: 

• Jitter under 30 ms is generally considered acceptable for maintaining good voice quality. 
• ExpertConnect uses jitter buffers to smooth out variations in packet arrival times. These buffers can be configured as: 
  • Small: ~20 ms buffer 
  • Medium: ~40 ms buffer 
  • Large: ~60 ms buffer (default for conferences)

          Currently we don't set these values exclusively and rely on the implementation defaults.

• Excessive jitter (e.g. >60 ms) can cause noticeable audio artifacts or delays, especially if the jitter buffer grows too large (e.g. >250 ms), which can introduce latency perceived by users. For best results, aim to keep jitter below 30 ms and ensure your network is stable and optimized for real-time traffic. 

Packet Loss

Our partner telephony provider recommends keeping packet loss below 1% for optimal voice call quality. Even small amounts of packet loss can lead to choppy audio, dropped words, or degraded call experiences, especially in real-time communication like VoIP. 

Best Practices to Minimize Packet Loss: 

• Use a wired connection instead of Wi-Fi when possible. 
• Enable Quality of Service (QoS) on your router to prioritize VoIP traffic. 
• Avoid network congestion by segmenting voice traffic (e.g. using VLANs). 
• Ensure your firewall and NAT settings are properly configured to support UDP traffic. 

Key Metrics Covered

• Bandwidth: Minimum ~100 kbps per concurrent call (upstream and downstream)
• Latency: Ideally under 150ms round-trip 
• Jitter: Should be less than 30 ms
• Packet Loss: 0% for best quality
• Firewall / NAT: Allow UPD ports 10000-60000

Call Forwarding from PBX Phone systems

• ExpertConnect calling is designed to function optimally when the customer dials the ExpertConnect VoiceHub number directly, initiating the workflow without external interference. 
• In scenarios where a PBX system or third-party phone infrastructure is position in front of the calling workflow through a call forward, additional latency may be introduced during the call handover process. 
• Since there is no native integration between ExpertConnect and external phone systems, forwarded calls are routed back through the public telephone network before reaching the VoiceHub number. This detour can impact the speed and reliability of the workflow, in which ExpertConnect will have no visibility to these factors. 

Firewall & NAT

• Details on specific firewall and secure network configuration can be found in this help article, which is kept up-to-date by the engineering team. 
• Whitelist our partner telephony provider and ExpertConnect product’s signaling and media servers.

Devices & Hardware

Headset Considerations: 

• These considerations apply to both desktop (PC or Mac) and mobile devices. 
• On desktop platforms, ExpertConnect is accessed via a web browser. This limits support for native headset functionalities and action button integrations, as such features are constrained by browser capabilities. 
• ExpertConnect currently carries QA testing on latest versions of Chrome and Edge browsers. 
• On mobile devices, the ExpertConnect app supports basic adio input/output routing but does not integrate with native headset features or action buttons. 
• These recommendations are not intended to endorse or require the use of any specific device or headset.

Mobile Device Considerations:

• Low-end or older devices may struggle with real-time audio encoding/decoding, especially during multitasking.
• Ensure the device is not thermally throttled or running low on battery, as this can
  impact audio performance.
• The ExpertConnect team currently tests calling functionality using a limited set of devices as part of the QA process prior to releasing mobile app updates to production. 
• At this time, we do not provide specific device recommendations. 
• Users are encouraged to follow the general device and hardware guidelines outlined above to ensure the best possible calling experience.

Brower Compatibility for WebRTC: 

• WebRTC performance and codec support vary across browsers. Outdated versions may lack critical updates. 

Software and App Configuration

Notes: 

• ExpertConnect releases updated versions of its mobile apps to the Play Store and App store on a bi-weekly schedule. Details about each release are provided in the Changelog and Viva Engage. 
• A common issue encountered on the web platform involves browser extensions that interfere with the rendering of ExpertConnect or disrupt calling functionality. These typically include: - Audio-related extensions - Ad blockers - Internet filtering tools - Network proxies or VPNs - Unverified or unknown browser extensions. To ensure optimal performance, users are advised to disable such extensions or test in an incognito/private browsing window to validate if any extensions are hindering the calling experience. 

Additional Resources: 

• Firewall & Secure Networks - ExpertConnect
• Troubleshoot Proxy Settings - ExpertConnect
• Troubleshooting Voice Issues (Jitter, Latency and Static) - ExpertConnect