The purpose of this article is to provide information to IT Admins and Network Engineers on the network requirements for Outreach Voice.
For more information regarding Outreach user best practices, refer to the Outreach Voice Best Practices article.
Note: Outreach uses webRTC for computer-based calls which makes performance dependent on the connection established between your network and Outreach servers.
- Minimum bandwidth: Outreach suggests a minimum of 100 kbps symmetrical connection for each concurrent call.
- To test network bandwidth, visit the Internet Speed Test web page.
- Recommended bandwidth: Outreach recommends an optimal bandwidth of 300-500 kbps for each concurrent call to handle the network requirements of reps multitasking while on a call.
Your firewall should allow both inbound and outbound traffic over Outreach ports and IPs to enable optimal function of WebRTC.
Signalling - 443 TCP
Media - 10000-20000 UDP
Only white-list the IPs which correspond to the region your office is located in. The following table is a complete list of IPs that may be used.
|Location||Media Server IP Address Range||CIDR Notation|
126.96.36.199 - 188.8.131.52
184.108.40.206 - 220.127.116.11
18.104.22.168 - 22.214.171.124
126.96.36.199 - 188.8.131.52
184.108.40.206 - 220.127.116.11, 18.104.22.168 - 22.214.171.124
126.96.36.199 - 188.8.131.52
184.108.40.206 - 220.127.116.11
18.104.22.168 - 22.214.171.124
126.96.36.199 - 188.8.131.52
184.108.40.206 - 220.127.116.11
18.104.22.168 - 22.214.171.124
|United States - East Coast (Virginia)||126.96.36.199 - 188.8.131.52 , 184.108.40.206 - 220.127.116.11||18.104.22.168/23, 22.214.171.124/23|
Reducing Jitter and packet loss
Jitter, and packet loss are two of the biggest contributors to voice quality issues in any computer-based dialing environment.
- Jitter: When packets arrive in a different order compared to when they were sent. The main symptom is robotic sounding audio..
- Packet loss: Certain networks such as WiFi are prone to packet loss with the primary symptom being choppy audio.
Outreach recommend the following best practice to avoid Jitter and Packet loss:
- Equip your reps with Ethernet when possible.
- Reduce packet conflicts on WiFi by reducing the number of devices operating on the same channel and avoid large data file transfers over the same WiFi environment with voice.
- Configure your router/firewall with QoS rules to prioritize the traffic on the ports 443 TCP and 10000-20000 UDP mentioned above.
- Configure your router with QoS rule to prioritize the traffic based on the IP ranges mentioned above.
- If using managed switches or certain routers/firewalls which support CoS, mark all Outreach Voice traffic with a DSCP of 46 (EF).
- Avoid bufferbloat. Bufferbloat builds up large queues that causes noticeable latency and bursts of jitter on a computer-based call.
- Outreach recommends configuring your router with a low buffer size around 100 ms or less.
Latency symptoms include call delays or people talking on top of each other. Callers typically start to notice the effect of latency once it breaches 250 ms for a “mouth to ear” trip, and above ~600 ms the experience is unusable.
The codec algorithmic time, the jitter buffer, and the traversal time between Outreach servers to your network will always introduce some level of latency. The object is to minimize it and keep the total trip time below 300-400 ms for computer-based calls.
Strategies to minimize Latency:
- Some lower bandwidth fixed internet connections can often have a higher latency. If possible, upgrade your internet connectivity.
- LTE (mobile 4G Data) and other cellular data solutions can often have high latency.
- Configure QoS (as mentioned above).
If your router includes SIP Application Level Gateway (ALG) function or Stateful Packet Inspection (SPI), disable both these functions.