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SipChannelReinvite Method (String, String, UInt32, String)

Issues a reinvite for the current call. This allows you to redirect the RTP traffic to a different address and is used to reduce latency in the audio stream.

Namespace:  VoiceElements.Client
Assembly:  VoiceElementsClient (in VoiceElementsClient.dll) Version: 8.6.1.1
Syntax
public bool Reinvite(
	string rtpCodec,
	string rtpAddress,
	uint rtpPort,
	string rtpSdp
)

Parameters

rtpCodec
Type: SystemString
The audio codec name to use with the reinvite.
rtpAddress
Type: SystemString
The address of the remote target endpoint.
rtpPort
Type: SystemUInt32
The port of the remote endpoint.
rtpSdp
Type: SystemString
The SDP (Session Description Protocol) of the first leg of the call.

Return Value

Type: Boolean
True for successful, false if failed.
Remarks
Documentation In Development

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See Also