RTP Video

The RTP Video tab is available when you select rtpvideo protocol (if your system has VoLTE license) when defining RTP Traffic (Gm | Media | RTP Traffic | DMF > Protocol - rtpvideo).

The rtpvideo protocol supports H.265, H.264, H.263 and VP8 Codecs. The DMF configuration includes a subset of the parameters required to accomplish RTP Video:

Originating Streaming mode: Bi-Directional, Tx Only, Rx Only
Terminating Streaming mode: Bi-Directional, Tx Only, Rx Only
Use same settings in both directions
Video MP4 file to play
Codec type and supplementary parameters (Payload Type, etc…)
Payload Type (see RFC-3551)
Jitter buffer size (in milliseconds)
Max Network delay (in milliseconds)

Originating Streaming

Terminating Streaming

Supports three modes of streaming video exchanged between Terminate and Originate ends of SIP Session/Call:

Bi-Directional (Full-Duplex) mode (Default)
Tx Only mode
Rx Only mode

Use Same Settings in both Directions (Available when Bi-Directional is selected)

The Bi-Directional mode assumes that after a SIP Session is established, the Terminate side always starts RTP data transmission that reflects the fact the callee speaks first. (Technically SIP ACK received by Terminate side almost guarantees that the Originate side is ready to accept RTP data.

NOTES:

The GUI Tool tip will suggest options for selecting Originator/Terminator mode. However, the user has the flexibility to select whatever mode they prefer.
If originating stream is Bi-Directional, then the tool tip will suggest Bi-Directional for terminating stream. This will only be a default behavior, but is entirely dependent on the Landslide user discretion.
If the originating stream is Tx only, then the tool tip will suggest Rx only for Terminating stream.
If the originating stream is Rx only, then the tool tip will suggest Tx only for Terminating stream.
No tool tip will be applied for Terminating stream.
Default behavior for both originator and terminator streams will be Bi-Directional.

Bi-Directional

Select to indicate two flows, Tx and Rx traffic flows.

Bi-Directional (Full-Duplex) mode simulates both-speak-both-listen scenario. In this mode both sides simultaneously transmit and receive encoded RTP streams.

RTP packet transmission starts when:

The Terminate end begins to transmit upon receiving SIP ACK.
The Originate side begins to transmit upon receiving the 1st RTP packet from Terminate.

NOTE: The encoded wav file (RTP Stream) transmits repeatedly during the SIP session (Call Hold time).

SIP Session stops gracefully as follows:

Current RTP Stream transmission cycle has finished AND
Current RTP Stream receive cycle has finished (all RTP packets within the Stream received ) OR
The last RTP packet arriving time expires (that is, if some of RTP packets within the Stream lost)

For example, if Call Hold Time = 10, File size = 8 seconds, and Maximum Network Delay = 2 seconds then in the worst case actual call hold time will be 8 + 8 + 2 = 18 seconds.

Tx Only

The Tx only mode simulates talk-only scenario. Both Originate and Terminate side of SIP session transmit encoded WAV file(s). The encoded wav file (RTP Stream) is either transmitting repeatedly during the whole SIP session (Call Hold time) or a number of times depending on rtpvoice DMF configuration.

Rx Only

The Rx Only Mode simulates listen-only scenario. Both Originate and Terminate side of SIP session receive encoded WAV file(s). (The incoming packet flow is analyzed and the RTP Rx Statistics are updated: total received, out-of-order, late arrival, duplicate, lost, interval, arrival jitter, and so on).

Use same settings in both directions

Both Originate and Terminate side of SIP session receive encoded WAV file(s).

Tcl Parameter : Direction

Jitter Buffer Size

The Jitter Buffer Size is used to determine RTP Packets Late Arrival. The RTP packets delayed more than the time entered in Jitter Buffer Size and less than the Max Network Delay are considered as late arrival.

Range: 0 to 65535

Default: 200 milliseconds.

Tcl Parameter : JitterBufferSize

Max Network Delay (ms)

Specifies the maximum time an RTP packet can take to travel from source to destination. If a packet does not arrive before the specified delay time, it considered as lost (RTP Packets Lost). All the packets arrived after Max Network Delay is discarded.

The value of Max Network Delay (ms) should be at least 20 milliseconds greater than Jitter Buffer size.

Range: 80 to 65535

Default: 1500

Tcl Parameter : MaxNetworkDelay

Codec Configuration

Select direction of the traffic flow, packet size and interval.

Codec

Support H.265 , H.264 , H.263 ,VP8

Tcl Parameter : Codec

Codec Parameters:

Payload Type

Payload type (PT) identifies the format of the RTP payload and determines its interpretation.

Range: 96-127.

Default: 96.

NOTE: This field value must be unique through "rtpvoice" and "rtpvideo" DMFs for all the test cases configured for any given test session.

Tcl Parameter : PayloadType

Call Configuration

(Only applies to L4-7 Data Traffic uses)

Only applicable when using configuration via L4-7 Data Traffic.

Call Hold Time (s)

Indicates the length of call.

Option: 1 - 65535

Default: 20 (s)

Tcl Parameter : CallHoldTime

Call Pending Time (s)

Indicates the idle time.

Option: 1 - 65535

Default: 1000

Tcl Parameter : CallPendingTime

Media TDF (.WAV)

The RTP Video DMF uses a TDF to select a .mp4 (recommended for H.263 and H.264 codecs), .webm (recommended for VP8 codec) and .mov (recommended for H.265 codec). The Video RTP packet stream is created from the .extension file.

You may also import your own .extension (.mp4, .webm, .mov) files into the TDF library.

Packet Size (bytes) - Enter packet size, Range from 100 to 1470. Default = 1024. Maximum packet size for the given direction is 1470.

NOTEs:

The maximum length of the file depends on Test Server’s memory and number of UE configured in a test as RTP stream is assigned individually per UE.
If codec and .extension file format do not match, then the file cannot be encoded and an encoding error is reported.

Document Number

Title

RFC 6386 - VP8 Data Format and Decoding Guide

ISO/IEC 14496-10. Information technology – coding of audio-visual objects - Part 10: Advanced Video Coding

RFC 7741 – RTP Payload Format for VP8 Video

RFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control

RFC 3550 - RTP: A Transport Protocol for Real-Time Applications

RFC 4566 - SDP: Session Description Protocol

3GPP TS 26.234 - Streaming Service (PSS); Protocols and codecs

ERD-PEVQ QoS Measurements for VoLTE on LS

RFC 7798 - RTP Payload Format for High Efficiency Video Coding (HEVC)