Keywords: asterisk

# Additional restrictions and caveats

One restriction that has not been spelled out so far but that has been hinted at is the presence of a bridge. The term bridge in this sense means two channels exchanging audio with one another. A bridge is required because use of a jitter buffer is a prerequisite for using PLC, and a jitter buffer is only used when bridging two channels. This means that one-legged calls, (e.g. calls to voicemail, to an IVR, to an extension that just plays back audio) will not use PLC. In addition, MeetMe and ConfBridge calls will not use PLC. It should be obvious, but it bears mentioning, that PLC cannot be used when using a technology's native bridging functionality. For instance, if two SIP channels can exchange RTP directly, then Asterisk will never be able to process the audio in the first place. Since translation of audio is a requirement for using PLC, and translation will not allow for a native bridge to be created, this is something that is not likely to be an issue, though. Since a jitter buffer is a requirement in order to use PLC, it should be noted that simply enabling the jitter buffer via the `jbenable` option may not be enough. For instance, if bridging two SIP channels together, the default behavior will not be to enable jitter buffers on either channel. The rationale is that the jitter will be handled at the endpoints to which Asterisk is sending the audio. In order to ensure that a jitter buffer is used in all cases, one must enable the `jbforce` option for channel types on which PLC is desired.