• RTP DMF 1                               Enable SRTP

• Type

• Role

• Rating Group

• Service ID

• RTP DMF 2                               Enable SRTP

• Type

• Role

• Rating Group

• Service ID

• RTP DMF 3                               Enable SRTP

• Type

• Role

• Rating Group

• Service ID

• RTP Traffic Start Delay (ms)
• Report Confidence Intervals
• Fireball

• Check Interval (ms)
• Number of UEs Performing RTCP
  • Selection Mode
    • Continuous
      • First UE
    • Random

• Transmit
  • Send Level (dBm)
• DTMF Tones
  • Time on (ms)
  • Interdigit Pause (ms)
  • Frequencies

• Send Level
• Detect Threshold
• Ringback Frequencies
• Ringback Times
• UEs Performing Call Progress

Enable RFC2833/RFC4733 - Support for Dual Tone Multi-Frequency (DTMF) on VoLTE.

• Event Type
• DTMF

Enable Audio Recording

• UE to Record
• Recording Time (s)

RTP DMF 1, 2 and 3

• Type

• Role

• Rating Group

• Service ID

Select the type, Role, Rating Group, and Service ID associated with the RTP traffic.

Select to Enable SRTP. A new tab SRTP will be enabled to enter Secure Real-Time Transport Protocol for a media file. Available for DMF 1, 2 and 3.

• Type — Indicates the type of media sent with DMF 1, DMF 2 and DMF 3. Select the relevant media type: Audio or (Video or RTT) (Real Time Text). Currently can only support Audio and either Video or RTT - Not all three. When an RTT DMF Type is selected the RTT fields are displayed instead of Video.  

• Role — Select the role of RTP DMF: Default, Server, Client.

• Rating Group — Indicates the group of services that share the same cost/rating type. A Rating Group can include one service or group of services that share the same cost and rating type — HTTP and email services could be defined in one Rating Group, for example, while multimedia services are defined in another.

• Service ID — Indicates the Service ID for which the credit is granted. The server can choose whether to grant credit for individual services within the same Rating Group or grant credit for the entire Rating Group. In the latter case, all services within a group can draw from the same credit pool.

Available only if there is at least one DMF included. Enter the required delay.

Default: 1000

Tcl Parameter: RtpTrafficStartDelay

Available in AMF Nodal, MME Nodal, and IP Application Node when Enable Mobile to Mobile is enabled on the WebRTC client.

Enable to report the confidence intervals on L57- Voice / Video.  

Tcl Parameter: RtpConfidenceStatsEn

Select to send RTP data at a faster rate. Simultaneous RTP Voice and Video are supported. This is a licensed feature.

Additional details in Fireball Data Message Flows. See this table for supported options.

Available in IP Application Node, MME Nodal, PGW Nodal and SGW Nodal test cases.

Tcl Parameter: VolteFireballEn

See Data Message Flow (Data Traffic Tab > Data Message Flows) to create RTP DMFs for VoLTE testing.

See DMF | RTP Voice for using rtpvoice protocol.

• Enable RTCP  — Enables the RTCP (Real-Time Transport Control Protocol - defined in RFC 3550) Pane which allows for capture of statistical and control data and works hand in hand with RTP that delivers the data. Additional OMs can be found in the RTP Voice and RTP Video Measurements. RFC 3661 - RTP Control Protocol Extended Reports (RTCP XR) and RFC 7266 - RTCP XR Blocks for Mean Opinion Score (MOS) Metric Reporting.

• Check Interval (ms) — Indicate the interval in milliseconds that the RTCP reports are sent  out. Range: 100 - 4294967295. Default: 5000.

• Number of UEs Performing RTCP - Enter the number of UEs performing RTCP. Because RTCP is time consuming, enter the number of UEs performing RTCP operations either Continuous range or pick a certain number of UEs randomly. If Continuous is selected, enter the First UE . By restricting the range we allow for high capacity RTP configurations in Fireball mode with RTCP.

Select Tones Transmit Send Level (dbm).

Options: - 4, - 7, - 10, - 13, - 16, - 19, - 22, -25

DTMF Tones.

• Enter Time on in milliseconds. Range: 75 to 3600000. Default : 150

• Enter Interdigit Pause in milliseconds. Range: 75 to 3600000. Default : 150

• Select Frequencies table:

Available if Enable RTP Traffic is selected, Functionality Pattern = All Originate or TOTO or OTOT and DMF = rtpvoice (on Nodal test cases).

Available if Mode = SIP Endpoint, Enable RTP Traffic is selected, Final Response Delay is selected, Functionality Pattern = All Terminate or TOTO or OTOT and DMF = rtpvoice (on IMS Node test case).

To make Ringtone detection possible go to Test Server Configuration and select Reserve Resources for : Digital Signal Processing (mutually exclusive with POLQA/VMAF).

Select to add support for ringback tone detection during SIP based MO call setup. SIP has its own embedded mechanism to notify Caller that the Callee is alerting: “180 – Ringing” message. An alternative way is to send ringback tone over early media. Early media is a data path that may have to be established between Caller side and Session Boarder Controller (SBC) or so after sending INVITE and receiving a final response (200 OK on INVITE or 4xx/5xx/6xx). The diagrams below show both ringing models: SIP 180 Ringing and Ringback Tone over early media.

Two Ringing models: SIP 180 (Left) and Ringback tone (Right)

Tcl Parameter : CptEn

Select the Send Level in dBm.

Default = -7

Range: -4, -7, -10, -13, -16, -19, -22 , -25

Tcl Parameter : CptSendLevel

Select the Detect Threshold in dBm.

Default = -23

Range: -11, -17, -23, -29

Tcl Parameter : CptDetectThreshold

We assume that ringback tone is consists of just two frequencies (f1 and f2),  More complex ringback tone, like music phrase or a voice announcement are not supported at this time.

Enter the f1 and f2 Ringback Frequencies (Hz).

Default for f1 = 440

Default for f2 = 480

Range: 50 to 3400

Tcl Parameter : RingbackF1

Tcl Parameter : RingbackF2

Enter the Ringback On and Off times is ms.

Enter the On and Off Ringback Times (ms).

Default for On = 2000

Default for Off = 4000

Range: 1000 to 10000

Tcl Parameter : RingbackOn

Tcl Parameter : RingbackOff

Enter the First UE and the total Number of UEs performing Call Progress.

Default for First UE = 1

Default for Number of UEs = 1

Range: 1000 to 10000

Tcl Parameter : CptFirstUe

Tcl Parameter : CptNumUes

Enable RFC2833/RFC4733 - Enable to support Dual Tone Multi-Frequency (DTMF).

Tcl Parameter : Rfc2833Rfc4733En

Event Type: DTMF -  Enable to VoLTE to emulate Dual Tone Multi-Frequency (DTMF) digit tone as audio encoded signal (G.711) or a Name Telephone Event (NTE) based on RFC2833/RFC4733.

DTMF are signals/tones that are sent when a user presses a telephone's touch keys. They are numbers 0-9, "*", and "#" keys on a standard telephone. DTMF tones or signals are used to access voice mail or navigate IVR.

RFC2833 describes the technique which uses a separate type of RTP payload, called Name Telephony Event (NTE), to carry DTMF digits. NTE are carried as part of the audio stream and must use the same sequence number and timestamps base as the regular audio channel to simplify the generation of audio waveforms at the gateway. Figure 1 show the payload format for NTE.

DTMF tones are delivered either in-band of out-of-band via SIP or RTP signaling messages. Two delivery options:

• RFC2833 - Sending DTMF information in the RTP stream as a named event (NTE).
• Inband - Sending DTMF information as an audio encode signal.

DTMF digits events are carried as part of the audio stream, and must use the same sequence number and timestamp base as the regular audio channel. The first packet for an event MUST have the M bit set.  The final packet for an event MUST have the E bit set. The update packets MUST have the same RTP timestamp value as the initial packet for the event, but the duration MUST be increased to reflect the total cumulative duration since the beginning of the event.

Tcl Parameter : DtmfEventEn

• Enable Send  — Enables the Digits pane which allows for capture of the Digit to be sent and the Duration.

• Payload Number — Indicate the Payload number. Range: 96 - 127. Default: 101.

• Alternate Payload Number — Enable to add an Alternate Payload number. Range: 96 - 127. Default: 101. Both the Payload number and the Alternate Payload number will be included in the SIP INVITE SDP. But on the terminator side, the Alternate Payload number will be used for telephone-event only if the Payload number is not included in the SIP INVITE message that it receives.

• Volume (dBm0) — Indicate the power level of the tone, expressed in dBm0 after dropping the sign. Range: 0 to -63 dBm0. Default: 8 dBm0.

Digits/Duration — Digits to be sent over DTMF. "None" will indicate Delay Time. As of release 17.2, this field will be disabled when using SIP Scripts. See SIP Actions (Media_DtmfStart, Media_pause and Media_resume). DTMF digits and timing added as input in Media_DtmfStart (Sip Action). Media_Pause / Media_Resume actions used to handle timing.

Digits to be send over DTMF, “None” will indicate delay time.

Digits Valid value: None, 0 – 9, *, #, A – D

Duration Range: 1 – 65535 milliseconds

Default value: None

Example configuration:

Measurements: Om counter for each individual event is need to accuracy measure/detect DMTF digits send or received.

Digit 0 through 9 – send and receive Om counter.

Digit * – send and receive Om counter.

Digit # – send and receive Om counter.

Digit A through Z – send and receive Om counter.

Available for MME Nodal and AMF Nodal. This option is enabled based on any of the selected DMFs being "rtpvoice" protocol.

It allows Landslide to record audio for 1 UE for the duration specified in GUI or Call is disconnected. The 1st call is recorded if there are multiple calls between A and B. Allows for audio recording without enabling POLQA.

The recording will be converted from RTP packets to Wav file and then exported to TAS as tc0_announcement.tar.gz. This announcement folder can be extracted. The audio file recorded will be name as Audio_ts0_tc0_ue0_stream-1_dmf0. The audio file can be played using Audacity tool. 

Enter UE to Record.

Range : 1 to "Number of Subscribers"

Default : 1

Enter Recording Time (s) in seconds.

Range : 1 - 3600

Default : 120

Limitation/Caveat :

1. As of release, 21.4, the following Codec are not supported for Audio recording. 
   1. G.729 A
   2. G.729 AB
   3. G.722
   4. OPUS
2.  If recording is enabled for A, the incoming RTP packets are arranged in recorder according to received RTP packet's timestamp. Therefore if A have Multiple dialogues, as in case with 3-way call (A calling B and C), recording data will be overridden if time stamp for incoming RTP packets for B and C are same.       
3. If there are multiple calls, 1st call is recorded. 
4. If the codec is changed during the recording, recorder will not work as expected.       

Tcl Parameter : AudioRecordingEn

Tcl Parameter : AudioRecordingFirstUe

Tcl Parameter : AudioRecordingTime