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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
)
Public Function Reinvite (
rtpCodec As String,
rtpAddress As String,
rtpPort As UInteger,
rtpSdp As String
) As Boolean
public:
bool Reinvite(
String^ rtpCodec,
String^ rtpAddress,
unsigned int rtpPort,
String^ rtpSdp
)
member Reinvite :
rtpCodec : string *
rtpAddress : string *
rtpPort : uint32 *
rtpSdp : string -> bool
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:
BooleanTrue for successful, false if failed.
Remarks Documentation In Development
We are constantly adding features and new capabilities to
The Voice Elements Platform. Therefore, our documentation of individual methods, properties and other elements
is constantly being developed and improved. If you would like to know more about this element, please help us perfect our documentation and help you develop your application by doing one or more of the following:
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See Also