All Jive Hosted VoIP handsets are considered network devices, and require an enterprise-grade network to function properly. Jive can recommend, configure, and sell network hardware as needed.
Routers and Switches
Only enterprise-grade routers and switches should be utilized when deploying Jive Hosted VoIP. Power over Ethernet (PoE) switches are highly recommended, as they eliminate the need for individual power adapters for each endpoint and also allow for centralized power redundancy.
The following protocols should be installed and available on the network:
Devices should receive an internal IP address assignment via Dynamic Host Configuration Protocol (DHCP).
Phones initiate a SIP session with Hosted PBX, NAT keep-alives need to be permitted to keep these sessions continuously open. Keep-alives sent on a 30 second interval.
In a converged network, Quality of Service (QoS) must be applied to prioritize voice and video traffic over all other traffic types.
DNS is used to resolve the URL for handset configuration files and time server. Most importantly, DNS is used to perform SRV lookups upon initiating every call.
Some Jive devices, such as Jive Voice Gateways and Jive HD Video units may require public IPs to function properly. Public IPs are not required (or recommended) for the Hosted VoIP handsets.
Be sure that the phones are using a reliable DNS server as a DNS failure can delay or prevent call setup. If a Jive Voice Gateway is deployed, best practice is to have this device provide both DHCP and DNS for the VoIP Handsets.
Firewalls should allow Jive Hosted VoIP handsets to access HTTP, HTTPS, and UDP traffic on the local network. Jive Hosted VoIP handsets must be allowed to both send and recieve TCP and UDP packets on arbitrary ports and to arbitrary IP addresses. Some network ports need to be opened manually.
Jive requires that firewalls allow the following activity for optimal functionality:
NAT keep-alives must be allowed every 30 seconds. (ports 5060 and 5061)
HTTP over port 80 must be enabled.
Multiple UDP connections must be allowed on ports 5060 and 5061.
Internally-initiated UDP requests must be allowed on ports 20,000-60,000 for audio (including non-T.38 faxing) and video.
Internally-initiated UDP requests must be allowed on ports 4,000-4,999 for T.38 Faxing.
UDP traffic must be allowed on port 123 for Network Time Protocol (NTP).
When a Jive Hosted VoIP Handset powers on, it initiates a SIP (UDP) session with Jive Core (in the cloud) on port 5060 or 5061. For service to function correctly, this session must remain open. Once the session has been established, Jive Core will send back NAT keep-alives every 30 seconds to keep that inbound connection active.
If your firewall drops these NAT keep-alives or 'prunes' NAT connections more aggressively than every 30 seconds, the handsets will not function properly. They will be able to call out, but will not receive inbound calls (inbound calls will go straight to voicemail). Accordingly, best practice is to ensure that any session expiration timers for these SIP sessions wait 90 seconds before closing a session.
Many routers and firewalls have SIP specific settings that manipulate how SIP traffic is handled. These settings almost always need to be turned off as they (somewhat ironically) will almost always break SIP.
Jive Voice Gateways are not strictly required for Jive Hosted VoIP to function properly. Jive Voice Gateways do, however, add functionality that can improve voice quality, increase network transparency, and allow for limited PSTN survivability.
The Jive Voice Gateway can be configured as your organization's gateway router or as a pass-through device for voice traffic only. Jive Communications handles the initial configuration for these devices, and prefers to maintain access to them for troubleshooting purposes.
Static IP Address
Jive Voice Gateways are premise-based components, configured and purchased through Jive Communications. For optimal results, a static IP assignment is required for the Jive Voice Gateway.
Quality of Service
In some networks, voice traffic can be routed through the Jive Voice Gateway. In these cases, the Jive Voice Gateway can be configured to provide QoS on voice traffic delivered to the WAN. If you are unsure of the capabilities of your current hardware, or are uncertain how QoS is to be configured, allowing Jive to deliver QoS via the Voice Gateway may be preferred.
Voice Quality and Network Monitoring
The Jive Voice Gateway allows you to monitor historical and active call quality. You can drill down to specific calls or times periods and see the Mean Opinion Score (MOS), latency, jitter, and packet loss of those calls.
Without the transparency the Jive Voice Gateway provides, troubleshooting call quality can be time consuming and difficult. The monitoring provided by the Voice Gateway decreases troubleshooting costs and speeds issue resolution.
In the event of a WAN failure, certain Jive Voice Gateways can be configured to automatically fail-over and route calls over the PSTN. When WAN connectivity is restored, calls will again be routed over the WAN.
To configure PSTN survivability, the Jive Voice Gateway is placed in transparent proxy mode. For this to work properly, the Jive Voice Gateway must have a public IP address to avoid NAT confusion between it and Jive Core.
The Jive Voice Gateways have the following hardware specifications:
1U (max, some are "desktop" form factor)
2 X 10/100 (ingress and egress)
T1 (optional), PSTN (optional)
Utilizing Virtual LANs to deploy Jive Hosted VoIP produces several network benefits. In general, it is beneficial to group network users and devices that share common traits. The primary mechanism for this is the creation of a Virtual LANs (or VLANs). Advantages include:
- Network segmentation. The network typically runs more efficiently, and some LAN issues will stay localized to VLANs.
- QoS. Can have a unique configuration for a specific VLAN.
- Policy-Based Routing. Unique policy-based routes can be configured differently on specific VLANs.
- Gateway Routing. VLANs can be used to route VoIP traffic through the Jive Voice Gateway.
There are three (3) methods to assign a Jive Hosted VoIP Handset to the desired VLAN:
Cisco Discovery Protocal.
Link Layer Discovery Protocol
Manually setting the VLAN ID on the handset
Jive provisions all Jive Hosted VoIP handsets with CDP and LLDP enabled by default.
If both CDP and LLDP-MED are enabled, the network policy for the VLAN is determined by the last policy set or changed with either one of the discovery modes. If both LLDP-MED and CDP are enabled, during startup, the phone sends both CDP and LLDP-MED PDUs at the same time.
Inconsistent configuration and behavior for network connectivity devices for CDP and LLDP-MED modes could result in an oscillating rebooting behavior for the phone due to switching to different VLANs.If the VLAN is not set via CDP and LLDP-MED, the VLAN ID that is configured manually is used. If the VLAN ID is not configured manually, no VLAN will be supported.
LLDP overrides CDP. CDP overrides Local FLASH which overrides DHCP VLAN Discovery.
Quality of Service
Quality of Service (QoS) refers to the capability of prioritizing certain types of traffic throughout the entire network. In the context of Jive Hosted VoIP, this term refers to the dynamic prioritization of Voice and Video over other traffic types. Particularly in scenarios where bandwidth is limited, properly configured QoS is crucial if maximum call quality is to be obtained. Properly implemented, QoS can reduce congestion, latency, and packet loss - all of which negatively impact call quality.
The QoS strategies discussed below pertain traffic inside the client LAN, and on the hand-off from the LAN to the WAN). Any QoS on the WAN connection itself must be delivered by the ISP. This is one of the primary motivators for purchasing Internet Access from Jive Communications (Jive Broadband), as QoS is applied across the WAN connection.
Several mechanisms for delivering QoS are available, and network administrators should use the option that best fits for network environment.
Physical Network Separation
A popular way to ensure network quality is to physically separate voice and data networks. This method involves using a dedicated WAN connection for voice only, and using separate WAN connections for data traffic. |
Balancing or Policy Based Routing
Another method for achieving QoS on the land is logical network separation. Networks can be separated into logical divisions or VLANs to separate voice from lower priority traffic. This traffic balancing, or policy-based routing, can allocate bandwidth dynamically based on volume, or statically by manual assignment.
If you have multiple WAN connections you can configure your network to route your voice VLAN out one WAN connection and all other VLANs over another. In this scenario, saturation of the 'data' WAN connection is irrelevant, as all voice traffic is routed over its own WAN connection.
Class of Service/DSCP
Routers and Gateways can be configured to honor Layer 3 DSCP values. Layer 2 802.1p/CoS values can also be used, though DSCP is preferred. As configured by Jive Communications, the VoIP handsets set a DSCP value in the header of each packet they generate, as shown in the following table:
Call Media (the actual voice conversation)
To enable QoS within the LAN and over the LAN-to-WAN hand-off, the network should be configured to prioritize traffic carrying those tags over all other traffic.
Implementing Downstream QoS (ingress) on your internet connection requires the cooperation of your ISP, as there is very little that can be done on a LAN to prevent saturation of the downstream link. This is a major reason why many Jive clients purchase their internet connection through Jive Communications. Jive is one of only a handful of providers that offer "end-to-end" QoS data solutions.