IMPORTANT

  • MEDIA should only be enabled with Jive’s assistance. If setup incorrectly, your phones could be down for up to 24 hours after it’s corrected.
  • MEDIA should never be enabled for the 0.0.0.0/0 network.

In general, there are two parts to a VoIP phone call:

  1. Media (RTP): The actual audio content.
  2. SIP Signaling: Also called the control channel; this is a relatively small stream of information that maintains detail/control about the call.

Most of time, both parts of the call route back through Jive’s datacenters. This consumes bandwidth which results in some delay (latency) even when calling someone inside the same physical office.

Under some circumstances, Jive is able to “release” the media such that the Media (RTP) stays as local to the endpoints as possible, including staying on the actual LAN (and not traversing back to Jive’s data centers at all).  There are a number of advantages to calls that are “released” in this manner; less WAN bandwidth consumption, reduced latency, etc.

Media Release Requirements

For media release to work between any two devices:

  1. The devices must sit behind the same Adtran router that is running in Transparent SIP Proxy mode (as other hardware platforms are certified, this page will be updated).
  2. The Adtran must be providing NAT/DHCP to the devices.
  3. The Adtran must be assigned a public (static) IP address.

Note: Forwarding an IP address will generally not work.mediaRelease

mediaRelease_multiple

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