AIS Managed VoIP Requirements
AIS Managed VoIP Network Requirements and Recommendations
8 minute read
This section provides AIS Managed VoIP customers with network requirements and recommendations to ensure that cloud-based unified communication services operate properly. For successful implementation, the network requirements must be followed without reservations, while recommendations are advised to be followed.
If your network is managed by AIS, your VoIP Project Manager will ensure that all requirements are met.
End-to-End Quality of Service Network Requirements
The requirements stated in Table 2 need to be satisfied for VoIP media traffic to get optimal call quality between extensions.
Network PropertyRequirementBandwidth | Each network connection in the end-to-end path must have a capacity in each direction that is larger than the maximum number of simultaneous calls plus capacity added for other types of non-real-time traffic and growth
Delay | lt 150 ms
Jitter | lt 30 ms
Packet Loss | lt 1
Network Readiness Assessment
The end-to-end quality of service requirements stated above can be validated by performing a network readiness assessment, which determines the quality of the local network and the Internet Service Provider network. Two types as network readiness assessments can be performed to assess the ability of the network to support AIS Managed VoIP communication services:
- Brief Network Assessment - This assessment leverages basic Capacity Test and VoIP Quality Test tools to test your Internet Connection for AIS Managed VoIP. These tools provide an impression of network capacity and quality in the outbound direction of an enterprise site to the AIS Managed VoIP AWS Region over a time interval of a few minutes.
- Comprehensive Network Assessment - In this case, a probe is installed at the enterprise site. By running this probe over a longer time interval (e.g. a full business week), a much better impression is obtained of the end-to-end quality and intermediate network hop quality in both directions of the call. Targeted network improvement recommendations can be provided based on this type of assessment.
The first type of assessment can be performed through self-service but provides minimal insights into the end-to-end QoS over time. The second type of network assessment, which is recommended to minimize the likelihood of user-perceived QoS issues, requires the involvement of AIS Engineers.
The requirements stated in the next sections must be implemented before a network assessment is performed so that any major network issues are already addressed.
Virtual LANs (VLANs) can be used as follows with AIS Managed VoIP
- Desk Phones and IP Speaker Phones - If VLANs are supported by network switches, then it is recommended to define a VLAN specifically for desk phones and IP speakerphones. This will keep VoIP traffic of these types of endpoints logically separate from data traffic and reduces broadcast domains. It also simplifies the management of these endpoints because their IP addresses are VLAN specific.
Small/Medium Businesses networks are mostly connected to cable provider or DSL ISP networks. These local networks may have lower quality equipment (such as all-in-one modems) than enterprise networks. Frequently, the users on such networks also use WiFi. The combination of these factors makes it more difficult to manage the end-to-end QoS for cloud communications services.
Many technologies exist to implement WANs, including internet, Ethernet Virtual Private Line, MPLS, and SD-WAN. Each type of network technology has its own way of supporting QoS. To ensure that the end-to-end QoS requirements and recommendations are met, it is required that every traversed WAN network segment must have sufficient quality.
Some network configurations are not supported/recommended for AIS Managed VoIP as they are known to cause continuous or intermittent voice quality issues (contributing to high latency, packet loss, or jitter).
The settings listed below may need to be adjusted on IP devices (Layer 3 Switches, Routers, Firewalls), and Ethernet switches, or be avoided.
Disabling functionality for the IP and higher layers can be limited to the Static IP Range of your AIS Managed VoIP instance by applying policy-based control.
OSI LayerSettingApplication | Session Initiation Protocol Application Layer Gateway (SIP ALG), Deep Packet Inspection (DPI), Application Layer Access Control, Stateful Packet Inspection (SPI), also called Dynamic Packet Filtering, Intrusion Detection/Intrusion Prevention System (IDS/IPS), WAN Acceleration
Transport | Port filtering
IP | Packet-by-packet load balancing across multiple Internet Service Providers links
Data Link | Auto-QoS, when used in combination with Polycom phones, Dynamic ARP Inspection
Physical | Energy Efficient Ethernet, Satellite network connections
Enabling these functions may result in intermittent call connectivity issues or excessive voice quality impairments (increased latency and jitter), specifically:
- For some of the functionality mentioned under Application Layer Functions, packet content may traverse a separate processing engine, which may result in the mentioned impairments. The impact may be minimal when using advanced networking devices but could be substantial for SMB devices.
- Enabling SIP ALG may cause signaling issues when desk phones and VoIP mobile apps are used simultaneously.
- IDS/IPS functions may limit packet streams to a certain bandwidth causing intermittent audio issues across multiple calls when the number of calls exceeds a certain volume. To reduce bandwidth, WAN accelerators use header compression to reduce traffic. For VoIP traffic, this can result in increased jitter.
- Port filtering, such as UDP flood protection, may limit bandwidth thereby causing intermittent voice quality issues when many simultaneous calls occur.
- Packet-by-packet load balancing may cause increased jitter and out-of-order packet arrival at the receiving media processor in the AIS Managed VoIP cloud instance. This may result in packet loss and intermittent or continuous voice quality issues, such as interruption of audio and SIP messaging in Session Border Controllers (SBC).
- Use of Auto-QoS may cause voice quality issues (such as distortions or incorrect volume levels) with older Polycom speakerphones and older versions of desk phones.
- Green Ethernet is used on switch ports to save energy by automatically turning them into low power mode after they have not passed traffic for some time. This may also cause intermittent signaling and media traffic issues.
- Satellite connections introduce delays much exceeding 150 ms in each direction and, depending on the quality of the satellite connection, may also cause excessive jitter and packet loss. It depends on end-user expectations whether this is acceptable.
AIS Managed VoIP uses Amazon Route 53 DNS Services for the following:
- Provisioning and firmware update services for desk and conference phones.
- Call servers.
- Presence status.
Endpoints access these services via DNS lookup to resolve a domain name into an IP address.
VoIP endpoints rely on a DNS service to resolve the call server domain name (e.g., voip.aislabs.com) obtained from the provisioning service to its corresponding call server address.
It is important that the domain name of the call server gets resolved to an IP address that is geographically close to the physical location of endpoints. Use of a single corporate DNS (e.g., country-wide or even a single global DNS) instead of a distributed DNS to resolve domain names to local IP addresses may result in longer paths to media servers, which adversely affects voice quality.
Network Address Translation/Port Address Translation functionality (generically referred to as NAT) is applied at the border between two networks to translate between address spaces or to prevent collision of IP address spaces. More specifically, a NAT function translates a source (IP address, port number) pair of outbound packets into a public source (IP address, port number) pair and maintains table entries corresponding to this translation to allow inbound response traffic to return to the proper host in the private network.
NAT is frequently implemented as part of a firewall functionality, but can also be implemented stand-alone.
For proper operation of the AIS Managed VoIP extensions, a minimum Network Address Translation time out needs to be configured. Cisco phones send a follow-up REGISTER refresh message every 4 minutes, Polycom phones every 5 minutes. As a consequence:
- NAT entry expiration timeout must be set to greater than 5 minutes to cover all extensions.
QoS Classification and Traffic Treatment Policies
AIS Managed VoIP traffic needs to be classified and treated properly in enterprise and service provider networks to ensure that end-to-end QoS requirements are met. In terms of QoS, VoIP and video impose the most severe constraints on the network because delay, packet loss, and jitter QoS requirements requirement need to be met. Signaling traffic has lower QoS requirements since real-time requirements do not apply and packets can be retransmitted when lost. Other types of service traffic, such as messaging and directory services, can be treated more like data traffic.
Ideally, COS tagging and DSCP marking values are used across the entire network between VoIP extensions and cloud-based AIS servers, and traffic is treated according to this classification, which is referred to as honoring the marking. However, in practice this is often not entirely possible because:
- Some network devices do not support sufficient QoS capabilities. Examples are low-end routers.
- COS values are often not managed in small networks.
- ISPs may change DSCP markings along the internet path, e.g. from DSCP 46 to 0.
- In large corporate enterprise networks, with sites connected to an MPLS or Metro-Ethernet network, a DSCP to COS mapping must be performed by the WAN network border devices.
- Some endpoint types do not mark COS/DCSP value yet
As communication methods become more fragmented, office phone use continues to decline but office phone system and service prices have increased in many cases. In designing a robust, reliable, VoIP platform, AIS selected the Asterisk VoIP server software, which is perhaps the most mature and widely deployed VoIP server. Hosting in Amazon Web Services with Geo-Redundant availability zones and connect to the public phone network using geo redundant sip trunks provided by Twilio allows for carrier grade reliability.
July 6, 2021