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SIP (Session Initiation Protocol)
5060 UDP (Desk phone) and 9002 TCP (Webrtc) Traffic - this is the call setup/signaling 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.
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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. *See “Ports and Protocols” above to make sure you have the proper ports open.
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.
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<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
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Common interruptions
SIP ALG - Present in many network device configurations as SIP ALG, SIP, VoIP, settings which enable SIP ALG are contrary to a successful network setup with multiple VoIP devices. Intermittent disconnection, audio drops, and the inability to register are common symptoms when SIP ALG is enabled. This must be shut off for optimal performance.
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.
Ensure UDP timeouts are greater than 240 seconds (phones register every 240 sec). If you’re seeing a sawtooth pattern in your latency graphs, or your phones are sometimes unable to be reached, the UDP timeout is probably incorrect.
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.