* The Telviva One infrastructure is new and Telviva One is not complete yet. But: to assist customers who need to accomodate work-from-home requirements due to the current corona virus pandemic, we are giving early access. We ask for your patience if there are teething troubles. The development team and every part of Connection Telecom will do their best to deliver a good result.
Telviva One is a web-based softphone application allowing users to make and receive calls from their web browsers using Telviva.
Telviva One allows users to leverage all the features offered on the Telviva platform, including call recording, conference calls and call transfers.
Telviva One fully integrates with various address books, Google Contacts and Telviva Shared contacts . The application is accessible for free at https://one.telviva.com . Telviva One turns your web browser phone into your home/office phone in an instant.
Your firewall should allow outgoing & incoming UDP to the public internet
We utilize Websocket connections so HTTPS / WebSocket / Secure RTP should be allowed
Local network conditions have the biggest impact on voice quality. Jitter, latency, and packet loss can be the biggest contributors to voice quality issues in any VoIP network.
The time it takes the RTP (media) packets to arrive at the destination
Causes media delivery delays, callers may speak over the top of each other.
Packets that don’t make it to the final destination
Causes gaps and cut-outs in media, callers may not hear the other side
Packets that arrive at the destination out of order
Cause a ‘robotic’ distortion effect in media, or packet loss when overrunning the jitter buffer
Latency: High latency can substantially degrade a caller’s experience. While there will always be some latency between the codec algorithm, the jitter buffer, and network traversal, the goal is to keep this to a minimum. Callers typically start to notice the effect of latency once it breaches 250ms, and find latency above ~600ms to be nearly unusable. Here are some strategies to minimize latency on your network:
Some lower bandwidth fixed internet connections can often have higher latency. If possible, upgrade your internet connectivity.
Stick to high-bandwidth connections. Mobile networks such as LTE (mobile 4G Data) can often have high latency.
Jitter: Packet loss, most frequently jitter-induced packet loss, can make a big impact on your VoIP call quality. Wi-Fi can be particularly bad for creating jitter. Here are some strategies to minimize jitter on your network:
Reduce packet conflicts on Wi-Fi by reducing the number of devices operating on the same channel.
Avoid large data file transfers over the same Wi-Fi environment concurrently with voice.
Avoid buffer bloat, which can result in high latency, and bursts of jitter. We recommend ensuring your router is configured with the low buffer size, as high jitter cannot be masked by a buffer without introducing artificial delay, and often choppy audio.Note: Not all routers allow for configuring buffer sizes, but some routers ship with defaults which are not optimized for real-time VoIP networks. Open-source routers, enterprise-grade routers and gamer-oriented routers are good candidates for providing the right configuration options and defaults.
If you have addressed the above issues and continue to have jitter related impact on your voice quality, you may consider configuring your router with QoS rules to prioritize traffic on the above media UDP ports. Given the large range of UDP ports, you should only do this with prior consideration to what other traffic may be flowing in that port range.
By following this guide, you can significantly improve quality of service for the wireless voice applications and reduce or eliminate dropped calls, choppy speech, fuzzy speech, buzzing, echoing, long pauses, one-way audio, and issues while roaming between access points.
3 key metrics for voice quality:
Network MOS – The Network Mean Opinion Score (MOS) is the network’s impact on the listening quality of the VoIP conversation. The score ranges from 1 to 5, with 1 being the poorest quality and 5 being the highest quality.
Packet Loss Rate – The packet loss rate is the percent of packets that are lost during transmission.
Interarrival Jitter – Interarrival jitter measures the variation in arrival times of packets being received in milliseconds (ms).
Below is a summary of the best practices to provide the best voice quality over wireless.
Perform a pre-install RF survey for overlapping 5 GHz voice-quality coverage with -67 dB signal strength in all areas. (Use Wifi Analyzer App)
If possible, create a new SSID dedicated to your voice over IP devices.
Set Authentication type to ‘Pre-shared key with WPA2’
Set WPA encryption mode to ‘WPA2 only’
Enable ‘5 GHz band only’.
Enable ‘Traffic shaping’ on the SSID to prioritize all voice traffic
SIP 5060 UDP / TCP – RTP 10000-20000 UDP – internal Network / UDP 65550 if Vibe goes via Firewall
Set DSCP to ’46 (EF – Expedited Forwarding, Voice)’ for RTP
What Telviva One needs to be able to access to log in and for a call to work:
Note the use of websockets – both to collect Telviva “events” and for the webrtc for calls. Some customers may have a web proxy running – some older web proxy software is not compatible with websockets even though they are a standard.
jt-hap01: address 18.104.22.168, tcp port 443: Used for access to the Telviva One backend services – straight https requests used, but also wss (secure websocket) connection for PBX events from Telviva.
tcp port 4433: WebRTC connections (this is an encrypted websocket connection carrying SIP packets to and from JSSIP on the client system)
tcp port 3478: STUN and TURN service, a requirement for WebRTC
TCP port 443 is the default port for secure http. It would be very unusual for that to be blocked; if it were blocked most websites wouldn’t work
TCP port 4433 is a fairly commonly used secondary port for secure http. But more likely that it is blocked. Might have to be unblocked on firewall.
TCP port 3478 is for STUN/TURN, a requirement for all webrtc and also used by modern SIP phones to help deal with NAT. Should be opened.
A customer using a WEB PROXY would need to check that it is compatible with the use of websockets. Websockets are a web/internet standard (https://tools.ietf.org/html/rfc6455) which should be supported.
To transfer a call using Telviva One, go to the Phone plugin in the bottom-right and dial a second number, or initiate another call from the Contacts tab. The original call will be placed on hold and offer options for transferring calls
Check your sound volume, mute settings and whether you are able to hear other sounds such as videos playing in the browser etc. If the sound is erratic, see the section about Requirements & Network above
Please check your microphone or headset first. You could use a tool such as sound recorder or your audio settings of your operating system to test sound input. If the sound is erratic, see the section about Requirements & Network above