Self Help
User Guide
Telviva One is an all-in-one communication platform which provides a cloud phone (fixed and mobile), video collaboration and team messaging. It is a web-based softphone application allowing users to make and receive calls from their web browsers using Telviva.
Telviva One is a web-based softphone application allowing users to make and receive calls from their web browsers using Telviva. It allows users to leverage all the features offered on the Telviva platform, including call recording, conference calls and call transfers, and offers access to the free call community between all customers.
Telviva One fully integrates with various address books, offering contacts and calendar integration.
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.
Telviva Service-specific features and benefits:
- Call transfers
- Speed dialling
- Call history
- Speakerphone, mute & hold
- Supports OPUS – G729 codec
- Meetings & conferences
- Team chat
- Mobile softphone
To access Telviva One, open the following link in your browser: https://one.telviva.com
We recommend using the Google Chrome browser, older versions of Internet Explorer is not supported.
Ports
Telviva One uses port 4433 and your network & firewall should allow traffic on this port.
In addition your device would have to have full access to https://one.telviva.com/api in order to access the required APIs
Routing
Your firewall should allow outgoing & incoming UDP to the public internet
We utilize Websocket connections so HTTPS / WebSocket / Secure RTP should be allowed
Wifi:
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.
| Latency | 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. |
| Packet loss | Packets that don’t make it to the final destination | Causes gaps and cut-outs in media, callers may not hear the other side |
| Jitter | 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.
Call quality
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
- network 197.155.248.128/25
- network 197.155.249.128/25
- network 197.155.250.128/25
- network 197.155.251.128/25
Set DSCP to ’46 (EF – Expedited Forwarding, Voice)’ for RTP
Video Meetings
Observations:
- 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.
Installation Guide
Logging in, out & Password Reset
You’ll see a screen like this to log you in:
Logging In
Password Reset
Logging Out
You can visit https://one.telviva.com/ from a desktop browser to log in on your own extension (login using your Telviva account details if necessary), or you can reset your password from this page
There are various permissions to be set and in addition you can allow the browser to serve notifications at the operating system level
| Setting permissions and notifications
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Microphone Permission
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| Permission Issues
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Link Google Account
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Making a call
Call History
Contacts page
To transfer a call using Telviva One, go to the web phone at 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.
Voicemail
Custom Voicemail Greeting
Voicemail settings
FAQ’s
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.
Please refer to the Requirements & Network section above.
Please follow the video instruction below
With 1Gb you should be able to make over 16.5 hours worth of calls.
Props to @scruzmusic on freesound.org for their public-domain audio samples.
You are welcome to contact Telviva Support on the following channels:
- Use the chat in the bottom right of this site.
- Call us on 0878 200 400
- email us on support@connection-telecom.com
- Use the form on the Contact Page