UCX Network Connectivity Requirements (including UCX Cloud)

Network Connectivity Requirements

Bandwidth Requirements

Every leg of a call that made to or from the UCX uses bandwidth in both directions during the call while it is on the IP network.  By default, the UCX uses G.711 encoding of the voice which requires approximately 100 kb/s of bandwidth per direction.


  1. If you are calling from an IP phone on your LAN to an analog or digital trunk, the bandwidth consumption for that call is about 100 kb/s second in each direction, but only on the 100Mb/s LAN connection.
  2. A remote user connecting to a SIP trunk on the UCX uses 100 kb/s each way across the WAN for the connection to the UCX and then another 100 kb/s each way across the WAN for the SIP Trunk portion of the call for a total of 200 kb/s each way for the call.
  3. An office worker calling another office worker in a UCX Cloud deployment will use 100 kb/s each way across the WAN to connect to the UCX Cloud. The UCX will then connect to the called worker using 100 kb/s each way, for a total of 200 kb/s each way for the call.
  4. A remote worker calling on a SIP Trunk on a UCX Cloud will not use any of the office WAN bandwidth, but will use 100 kb/sec each way on the home internet connection.

So the total bandwidth required for UCX IP calls will vary based on your network topology and the number of simultaneous calls your business will require. But remember that there are other services such as web browsing, streaming services, email and other corporate data usages that require the bandwidth as well, so it is important to assess the overall bandwidth requirements. E-MetroTel recommends planning for an 80% 

Latency Requirements

Latency is a measure of the time it takes for (voice) packets to reach the far end of a connection. For voice calls the higher the latency, the more difficult it is to communicate effectively over the connection, and may even cause calls to drop. E-MetroTel recommends that the end to end latency across a WAN connection to another site or to a UCX Cloud server less than 100 ms.

Jitter Requirements

Jitter is the amount of variation in the time it takes for packets to arrive at their destination, i.e. the inconsistency of the latency. Too much jitter can cause significant call quality issues. E-MetroTel recommends jitter be less than in the 20-30 ms range, and certainly never over 100 ms.

Packet Loss

While some degree of packet loss is tolerable in a Voice over IP call, packet loss over 3% can begin to significantly degrade the call quality.

Router Requirements

E-MetroTel does not recommend specific brands of routers. However, some of the router capabilities used to connect the office to the WAN need to be taken into consideration.

SIP ALGs (Application Layer Gateway) and Stateful Packet Inspection (SPI)

Many brands of routers offer some level of firewall capabilities that are intended to make intelligent decisions on which packets can pass from the WAN to the customer LAN. These capabilities attempt to determine where certain packets really should go once they are passed (or if they pass) through the firewall, and they don't necessarily make the correct choices. SIP ALGs will often result in one-way audio , no ringing, or dropped call. 

E-MetroTel requires that SIP ALGs be turned offFor more information refer to Unexpected call failures and registration problems.

Quality of Service

Many business class routers can be configured to prioritize certain packets over others as they pass from the high-bandwidth environment of the LAN to the narrower bandwidth WAN. In many cases this involves having the router look at a certain portion of the packets to determine if they have been marked with special Differentiated Services (DiffServ) codes. E-MetroTel marks VoIP related media packets with the Expedited Forwarding (EF) code by default, which is the industry standard. However, if your router requires some other value, refer to SIP Settings for an explanation of how this can be changed on the UCX.