Due to the nature of VOIP services overall, call quality can highly depend on the network speed, number of concurrent calls, the specific client device, etc. To help our customers optimize the quality of Outreach voice, we have compiled a list of necessary network settings as well as best practices for both Outreach users and org IT administrators (network engineers) when setting up Outreach Voice.
Outreach VOIP uses webRTC, making performance largely dependent on the connection established between your network and our servers. This section contains some network optimization that will improve the performance of Outreach Voice.
- Minimum bandwidth: We suggest you have a minimum of 100 kbps symmetrical connection for each concurrent call. You can use this link to test your existing network bandwidth.
- Recommended bandwidth: Typically we see reps engaging with other network activities while on the call (such as updating CRM record, navigating between different websites for research, etc.). If you engage in these network activities while calling prospects, we recommend 300-500kbps bandwidth for each concurrent calls.
Your firewall should allow both inbound and outbound traffic over our ports and IPs to ensure that our WebRTC client can optimally function.
Signalling - 443 TCP
Media - 10000-20000 UDP
Here is a complete list of IPs which may be used. Please note that you only need to whitelist the IPs which correspond to the region your office is located in.
The purple values in the table below will be available starting November 29, 2019. If you are whitelisting, we recommend you also whitelist the highlighted values.
The new United States - West Coast location will also be available starting November 29, 2019.
|Location||Media Server IP Address Range||CIDR Notation|
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
126.96.36.199 - 188.8.131.52
|United States - East Coast (Virginia)||184.108.40.206 - 220.127.116.11 , 18.104.22.168 - 22.214.171.124||126.96.36.199/23, 188.8.131.52/23|
|United States - West Coast (Oregon)||184.108.40.206 - 220.127.116.11||18.104.22.168/24|
Network Best Practices
This section contains some best practices/suggestions that will further improve the performance of Outreach Voice.
Reducing Jitter and packet loss
Jitter, latency and packet loss are the biggest contributors to voice quality issues in any VOIP 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 networks such as WIFI are prone to packet loss (also referred as a robotic sounding)
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. Avoid large data file transfers going over the same WiFi environment concurrently 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 in a VOIP call.
- We recommend ensuring your router is configured with a low buffer size. The perceived buffer size should be around 100ms or less.
Typical symptoms of latency are call delays or people talking on top of each other. Callers typically start to notice the effect of latency once it breaches 250ms for a “mouth to ear” trip, and above ~600ms the experience is unusable.
Note that there will always be some latency – the codec algorithmic time, the jitter buffer, and the traversal time between our servers to your network will always introduce some level of latency. The object is to minimize it and keep the total trip time below 300-400ms for VOIP 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) can often have high latency.
If your router includes SIP Application Level Gateway (ALG) function or Stateful Packet Inspection (SPI), disable both these functions.