Recommended Network Settings for VOIP
In This Article:
- Services To Disable
- Services To Enable
- Subnets To Whitelist and Ports To Open
- AVATAVA Public Subnets
- AVATAVA Required Ports
- Bandwidth Requirements
- Packet Measurements
- Packet Delay
- Packet Delay Variance
- Packet Loss
We recommend that a qualified network professional configure these advanced settings on end-user networking devices. The below settings are recommendations and should be implemented with careful consideration. AVATAVA is not responsible for any disruption to network services that may result in implementing these recommendations.
The following information, referenced in the AVATAVA Master Service Agreement (Section 5.4.1), is what AVATAVA considers to be "the minimum requirements" for operating VoIP over a network. AVATAVA does recommend the implementation of QoS (Quality of Service) mechanisms such as class-of-service and flow-control or the utilization of VLANs across your network.
Services To Disable
SIP ALG (Application Layer Gateway) functions such as SIP Transformations, SIP Application Helpers, SIP Normalization, etc. AV Client Enforcement on any IP assigned to an endpoint Content Filtering on any IP assigned to an endpoint
Services To Enable
Bandwidth Management/Traffic Shaping Default UDP session timeout to 300 seconds Consistent NAT (On Sonicwall devices) Inbound and outbound traffic on ports and subnets listed below DNS resolution on endpoints
Subnets To Whitelist and Ports To Open
IPv4
- 8.2.200.64/28
- 69.166.104.64/28
- 69.166.106.16/28
IPv6
- 2605:3ac0:100::1/56
AVATAVA Required Ports
Port Protocol Application 80 TCP HTTP 443 TCP HTTPS 2195 TCP SSL 5060 TCP and UDP SIP 5061 TCP and UDP SIPS 9002 TCP WebSockets 20000-27999 UDP RTP
Information regarding the application each port is for can be found in the knowledge base article: Ports Used By the AVATAVA Cloud Switch
Bandwidth Requirements
AVATAVA primarily utilizes the G.711 μ-law codec for VoIP traffic. Voice traffic is sampled at 8kHz frequency with 8-bit samples requiring 64kbit/s in both directions (the upload traffic representing your spoken audio and the download traffic representing the callers audio). To give plenty of overhead to the call (and accommodate for improvements in the G.711 protocol) we recommend 100kbit/s of allowance per concurrent call. So if 5 simultaneous calls are expected, a total of 500kbit/s should be available for VoIP calling to guarantee an error-free, high-quality call.
Packet Measurements
The following measurements are an integral part of the Mean Opinion Score (MOS) measured by the AVATAVA Softswitch. Calls should stay above a 4.0 measured MOS score to maintain a great audio experience. For more information regarding MOS scores, please review Mean Opinion Score (MOS)
Packet Delay
Packet Delay, often referred to simply as latency, refers to the amount of time it takes for a packet to be transmitted across a network from source to destination. For VoIP, high delay can be disruptive because it impedes the synchronicity of a conversation, often causing speakers to ‘step’ on one another. While a small amount of delay is unavoidable in Ethernet-based networks, problems can occur when the delay is not uniform from packet to packet or is too long. For acceptable voice quality, packet delay should be below 70 milliseconds (70ms) each way.
Packet Delay Variance
Packet Delay Variance, or jitter, is the variation in packet delay caused by the routing of Ethernet packets through the network. In general, higher levels of packet delay are more likely to occur on either slow or heavily congested links. Most VoIP endpoints have a jitter buffer to put packets back in order before being played to the listener, but these are often limited in size. For acceptable voice quality, packet delay should be below 10 milliseconds (10ms).
Packet Loss
Packet Loss is the percentage of packets transmitted from the source to the destination that "does not make it" due to factors on the Ethernet network. As packet loss increases, encoded audio is not being received at the destination which may cause odd audio "artifacts" including static, echo, and the total loss of audio in short segments. Packet loss should stay below 5% of 100 ~1500 byte packets to maintain acceptable audio quality.
