Client Workstation Requirements

To use the Sharpen platform as it is intended, we recommend the following specifications to provide a positive user experience while running with your other tools. Though the application may function while operating below these standards, we will focus our support on systems compliant with our recommendations.

Network Requirements

Ports and Protocols

It is important to make sure the following ports/protocols are free to communicate with our environment.

• 80 HTTP - Default port for web browser traffic

• 443 HTTPS - Default secure port for web browser traffic

• 5060 TCP/UDP Traffic -  SIP signalling traffic for IP Phones

• 8089 TCP - Used for establishing Websocket connections.

• 8090 TCP - Used for establishing Websocket connections.

• 9002 TCP - Used for establishing WebRTC connections.

• 10000-20000 UDP - Port range used for media transmission through WebRTC.

• Traffic allowed to and from

  • *.sharpencx.com

  • *.sharpen.cx

  • *.cx.shpn.co

  • *.sipvbx.com

  • *.fathomvoice.com

  • *.s3.amazonaws.com

  • *.yealink.com

  • *.ckeditor.com

• If whitelisting traffic , take note of the following tables

Prioritization

Use Quality of Service to maintain prioritization
Many devices support Quality of Service (QoS) tags to maintain traffic priority across the network. It is beneficial to tag your voice traffic with the appropriate tags, so it can be prioritized anywhere in the network in the event of a saturated link. This will help to prevent any audio issues caused by voice and data competing for the same bandwidth over your internet connection.

Use traffic shaping to offer voice traffic the necessary bandwidth
Due to potential contention of competing data on your network, it is important to ensure that your voice traffic has enough bandwidth to operate. As such, traffic shaping rules can be implemented to allow voice traffic to use additional bandwidth, or even limit other types of traffic to prioritize voice traffic.

QoS Settings

Protocols

SIP (Session Initiation Protocol)

• 5060 TCP/UDP Traffic -  this is the information about the call, such as phone 1 is calling phone 2 on server XYZ.

• 10000-20000 UDP Traffic - this is the Real-time Transfer Protocol (RTP) stream where actual packets of voice data are transmitted. This is the audio of the call.

While at rest, the phones only send 5060 UDP data as a ‘keep alive’ method for Network Address Translation (NAT), during this period there is no RTP traffic. Once a phone call is made and audio established, RTP traffic is sent from the phone to our servers.

WebRTC

WebRTC is an HTML5 specification which can be used to facilitate real-time media communications (video and audio) between browsers and other audio endpoints. The Sharpen Q phone built into the Sharpen Q application leverages WebRTC, allowing for seamless integration to the platform within the browser. The following are necessary for successful webRTC functionality.

• 80 HTTP - Default port for web browser traffic

• 443 HTTPS - Default secure port for web browser traffic

• 8089 TCP - Used for establishing Websocket connections.

• 8090 TCP - Used for establishing Websocket connections.

• 9002 TCP - Used for establishing Websocket connections.

• 10000-20000 UDP - Port range used for media transmission through WebRTC

Network performance

Uninterrupted, consistent network performance is required for a good experience with the Sharpen platform. Due to the inherent nature of Voice interactions to be real-time, we need consistency in the underlying network. Otherwise, users may experience dropped calls, choppy call quality, latency, or an overall slow experience.

Assessment

Run this basic speed test to obtain your download, upload, latency, and jitter results.

• <150ms latency to *.sipvbx.com (example: us1-webrtc-11.sipvbx.com for east coast US, or us2-webrtc-02.sipvbx.com for west coast US)

• Average latency variation < 30ms

  • High variation represents interruption to your connection. This may be a result of competing network traffic, or general hardware/network instability

  • While high latency on its own simply means delay, latency variation typically comes coupled with packet loss, which will mean dropped calls and/or choppy audio

• >10Mb/s internet connection

  • While voice itself is a fairly light operation, we recommend having enough bandwidth to handle all your operations. This value is more of a guideline, rather than a requirement. Most important is making sure your collection of tools, including Sharpen, have sufficient bandwidth.

  • The best way to determine bandwidth needs is to sample your tool set usage, and extrapolate from there.

  • Sharpen bandwidth utilization can vary widely based on how it is used some base-line examples of usage are as follows

    • Sharpen Q page load for single agent logged into 4 queues

      • ~350 KB transferred

      • ~7 MB page resources

    • 1 minute outbound call from Sharpen Q

      • ~125 KB transferred

      • ~1 MB page resources

    • Reporting/Insights (10 reports) page load

      • ~150 KB transferred

      • ~ 8 MB page resources

• < 1% packet loss

  • Voice requires basically no packet loss. If any packets are dropped, it will interrupt the audio stream. For this reason, packet loss directly influences the quality of a call. Enough packet loss will cause dropped calls

• < 30ms jitter

  • Jitter is the variation in delay of packets. Having high jitter will also cause poor call quality

Additional recommendations

• Sharpen strongly discourages the use of “combination” network equipment such as all-in-one modem/router devices. ISPs typically provide or rent these out to customers. They are known for having issues with VOIP traffic as well as having limited access to critical settings. If you are using a combination device and experiencing issues, the first step is to acquire a stand-alone modem and router. 

• If you can not acquire a stand-alone modem and router, we recommend reaching out to your ISP to see if they can place your combination device in “bridge” mode, then purchasing and connecting a 3rd party router to handle the local networking. 

• Allow SIP, UDP and RTP protocols - Sharpen uses these to send and receive traffic.

• Ensure UDP timeouts are greater than 240 seconds (phones register every 240 sec).

• Disable Stateful Packet Inspection (SPI) as it often flags VOIP traffic incorrectly. 

• Disable SIP ALG and/or SIP Transformations (this varies by router).

• Disable any VOIP specific functions that come pre-setup on your network equipment.

• Sharpen traffic over a VPN is discouraged due to likely latency and quality of service concerns.

• Sharpen traffic over an MPLS is discouraged due to potential inefficient route paths to voice resources.