Network & Connectivity Requirements

This article provides a detailed overview of the network and connectivity requirements essential for the RingDNA Communication Hub. It includes information about the servers, port ranges, and IP addresses that users must have access to.

Connectivity Overview

The RingDNA Communication hub offers two options for call connectivity.

Browser Calls

Call Forwarding

  • Uses the WebRTC protocol built in to Chrome for a seamless calling experience within the browser.
  • Audio is captured by the browser using a connected microphone.
  • Connection to carrier network is established over the internet.
  • Calls to/from RingDNA are forwarded from the carrier partner to the number specified.
  • Ideal for businesses with a strong PSTN backbone or mobile connectivity if office network can't support VOIP calling.

Bandwidth Requirements

Fast internet isn’t the same as consistent, reliable internet. Even if your connection has very high-bandwidth or speed, there could still be issues with jitter, packet loss, bufferbloat, or latency resulting in call quality issues.

The following table lists the network requirements to deliver acceptable audio quality through WebRTC with the OPUS codec:

Metric Recommendation

Latency (RTT) Less than 200ms
Jitter Less than 30ms
Packet Loss Less than 3%
Bandwidth

Recommended

Users engaged with other network activities while on the call (such as updating Salesforce or researching websites) should have at least 300-500kbps for each concurrent call.

Minimum

100kbps symmetrical connection for each concurrent call.

Network Tests  
WebRTC & Bandwidth https://networktest.twilio.com/
Packet Loss https://packetlosstest.com/

For more information see: Running Network Tests and Interpreting Results.

 

Firewall Configuration

Firewalls are used by network administrators to protect a private network by blocking or allowing traffic to and from internet destinations based on certain rules, such as traffic direction, protocol, and IP address.

Domain Connectivity Requirements

In order to use the RingDNA Communication hub, traffic must be allowed to and from the following domains.

*.ringdna.net, *.ringdna.com, *.pubnub.com, *.pndsn.com, *.pubnub.net and *.pubnubapi.com, *.launchdarkly.com

If access is not allowed for all domains listed above, the RingDNA Communication hub may not work as expected.

Carrier Connectivity Requirements

Your firewall should also allow outgoing TCP and UDP traffic to our carrier partner's media servers and signaling gateways, and allow return traffic in response.

Secure Media (ICE/STUN/SRTP) Edge Locations Protocol Source IP Source Port † Destination IP Ranges Destination Port Range
Global Media IP Range (Launching Oct 10 2023) UDP ANY ANY

168.86.128.0/18

10,000 - 60,000
sydney (au1) UDP ANY ANY 54.252.254.64 - 54.252.254.127
and
3.104.90.0 - 3.104.90.255
10,000 - 20,000
sao paulo (br1) UDP ANY ANY 177.71.206.192 - 177.71.206.255
and
18.228.249.0 - 18.228.249.255
10,000 - 20,000
dublin (ie1) UDP ANY ANY 54.171.127.192 - 54.171.127.255
and
52.215.127.0 - 52.215.127.255
10,000 - 20,000
frankfurt (de1) UDP ANY ANY 35.156.191.128 - 35.156.191.255
and
3.122.181.0 - 3.122.181.255
10,000 - 20,000
tokyo (jp1) UDP ANY ANY 54.65.63.192 - 54.65.63.255
and
3.122.80.0 - 3.122.80.255
10,000 - 20,000
singapore (sg1) UDP ANY ANY 54.169.127.128 - 54.169.127.191
and
3.1.77.0 - 3.1.77.255
10,000 - 20,000
ashburn (us1) UDP ANY ANY 54.172.60.0 - 54.172.61.255
and
34.203.250.0 - 34.203.251.255
10,000 - 20,000
umatilla (us2) UDP ANY ANY 54.244.51.0 - 54.244.51.255
and
54.244.51.0/24
1,024 - 65,535

roaming (gll)

UDP

ANY

ANY

All IP addresses listed above

10,000 - 65,535

Signaling Gateways Protocol Source IP Source Port † Destination Destination Port
Secure TLS to Carrier Partner TCP ANY ANY

chunderw-gll.twilio.com*

chunderw-vpc-gll.twilio.com*

voice-js.roaming.twilio.com

443

† The client will select any available port from the ephemeral range. On most machines, this means the port range 1,024 to 65,535

Global Low Latency (GLL) Requirements

By default, the ringDNA Communication Hub will resolve a hostname to the edge location with the least latency. In order for GLL to give accurate results, the intermediate DNS must:

  • Support RFC 7871 - Client Subnet in DNS Queries.
  • Reside in the same edge as the endpoint. For example, a user in the US configured with a VPN to Europe or configured with a DNS server that resides in Europe will result in connecting that user to an edge in Europe.

 

Salesforce Authentication & API Access

If your Salesforce Org restricts login access to a range of IPs, you will need to whitelist our application IPs in Salesforce. More information can be found in this troubleshooting article: Salesforce OAuth Failed: Ip Restricted

 

Jitter, Packet Loss and Latency Recommendations

Jitter, latency and packet loss are the biggest contributors to voice quality issues in any
voice network.

  • Jitter: When packets arrive in a different order compared to when they were sent. The main symptom is choppy audio quality.
  • Packet Loss: Certain network connections such as WiFi are prone to packet loss, which leads to “robotic sounding” audio.

We recommends the following to avoid jitter and packet loss:

  • Users should connect to the network using an ethernet cable whenever possible.
  • Reduce packet conflicts on Wifi by reducing the number of devices operating on the same channel or Wifi access points.
  • Avoid large data transfers or streams over the same WiFi environment concurrently with voice and schedule backups during off-hours.
  • Implement QoS rule to prioritize the traffic based on the IP ranges and internet addresses in this documentation.

A typical symptom of latency is audio delay or people talking over each other. Callers
typically start to notice the effects of latency once it breaches 250ms “mouth-to-ear” or roundtrip, and above ~600ms the experience is very poor.

Note that there will always be some latency – It takes time to encode the
audio and traverse the carrier’s networks. The goal is to minimize the total trip time below 300-400ms for voice calls.

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