## FsRtpSession FsRtpSession — A RTP session in a FsRtpConference |

## Properties

gint | no-rtcp-timeout | Read / Write
FsRtpHeaderExtensionGList * | rtp-header-extension-preferences | Read / Write
FsRtpHeaderExtensionGList * | rtp-header-extensions | Read
guint | send-bitrate | Read / Write
guint | ssrc | Read / Write
GObject * | internal-session | Read

## Types and Values

struct | FsRtpSession

## Object Hierarchy

        GObject
        ╰── FsSession
            ╰── FsRtpSession

## Description

This object represents one session, it is created by called `fs_conference_new_session()` on a FsRtpConference. It can be either Audio or Video. It also represents data send with one and only one SSRC (although if there is a SSRC collision, that SSRC may change).

### Codec profiles

It is possible to define "codec profiles", that is non-autodetected encoding and decoding pipelines for codecs. It is even possible to declare entirely new codecs using this method. To create a profile for a codec, add it to the codec-preferences with special optional parameters called "farstream-send-profile" and "farstream-recv-profile", these should contain gst-launch style descriptions of the encoding or decoding bin. As a special case, encoding profiles can have more than one unconnected source pad, all of these pads should produce application/x-rtp of some kind. The profile will be ignored if not ALL pads match currently negotiated codecs. Also, it is possible to declare profiles with only a decoding pipeline, you will only be able to receive from this codec, the encoding may be a secondary pad of some other codec.

### SRTP signature and encryption

To tell FsRtpSession to authenticate encrypt the media it is sending using SRTP, one must set the parameters using a GstStructure named "FarstreamSRTP" and passing it to `fs_session_set_encryption_parameters()`. The cipher, auth, and key must be specified:

#### FarstreamSRTP

`"rtp-cipher" and "rtcp-cipher"` | gchar * | Encryption algorithm Possible values: "null", "aes-128-icm" or "aes-256-icm"
`"cipher"` | gchar * | Default value for "rtp-cipher" and "rtcp-cipher" Possible values: "null", "aes-128-icm" or "aes-256-icm"
`"rtp-auth" and "rtcp-auth"` | gchar * | Authentication algorithm, can never be null Possible values: "hmac-sha1-32" or "hmac-sha1-80"
`"auth"` | gchar * | Default value for "rtp-auth" and "rtcp-auth" Possible values: "hmac-sha1-32" or "hmac-sha1-80"
`"key"` | GstBuffer | Size must be 30 if cipher is "aes-128-icm" and 46 if cipher is "aes-256-icm"

## Functions

## Types and Values

### struct FsRtpSession

    struct FsRtpSession;

## Property Details

### The `“no-rtcp-timeout”` property

      “no-rtcp-timeout”          gint

This is the time (in ms) after which data received without RTCP is attached the FsStream, this only works if there is only one FsStream. -1 will wait forever. 0 will not wait for RTCP and attach it immediataly to the FsStream and prohibit the creation of a second FsStream.

Flags: Read / Write

Allowed values: >= G_MAXULONG

Default value: 7000

### The `“rtp-header-extension-preferences”` property

      “rtp-header-extension-preferences” FsRtpHeaderExtensionGList *

GList of RTP Header extensions that are locally supported and desired by the application.

Flags: Read / Write

### The `“rtp-header-extensions”` property

      “rtp-header-extensions”    FsRtpHeaderExtensionGList *

GList of RTP Header extensions that have been negotiated and will be used when sending of receiving RTP packets.

Flags: Read

### The `“send-bitrate”` property

      “send-bitrate”             guint

The bitrate that the session will try to send at in bits/sec.

Flags: Read / Write

Default value: 0

### The `“ssrc”` property

      “ssrc”                     guint

This is the current SSRC used to send data (defaults to a random value).

Flags: Read / Write

Default value: 0

### The `“internal-session”` property

      “internal-session”         GObject *

Internal RTPSession object from rtpbin.

Flags: Read