Synthetic RTP sender

Gossipper can generate and transmit RTP streams without a media file. Silence frames are produced programmatically at the configured codec rate, which lets you load-test media paths, validate RTP reachability, and verify HEP/RTCP plumbing without preparing .wav or .raw fixtures.

There are two independent entry points:

Mode	Trigger	Use-case
Standalone CLI	-rtp_send flag	Send RTP without any SIP scenario
Scenario action	rtp_stream="synthetic,…"	Emit synthetic RTP inside an existing call flow

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Standalone CLI mode

Run Gossipper as a pure RTP generator by passing -rtp_send. The SIP scenario engine is bypassed entirely. You may prefix the same flag tail with gossipper sipp if you want the SIPp-oriented entry point; behavior is identical to the root command.

For thousands of parallel streams on one host, add -media_scale and -rtp_streams N (see docs/media-scale.md).

Required flag

Flag	Description
-rtp_send	Enable standalone RTP sender mode
-rtp_addr <host:port>	Destination UDP address

Optional flags

Flag	Default	Description
-rtp_codec <name>	PCMU/8000	Codec descriptor; sets PT, clock rate and packet size
-rtp_pt <n>	0	Override payload type number after codec params are applied (useful for dynamic PTs)
-rtp_freq <ms>	20	Packet interval in milliseconds
-rtp_dur <ms>	0	Total duration in milliseconds; 0 runs until interrupted
-rtp_ch <n>	1	Number of audio channels
-i <ip>	0.0.0.0	Local bind IP (shared with the SIP engine)

Supported codec descriptors

Descriptor	PT	Clock rate	Pkt interval	Payload/pkt
PCMU/8000	0	8 000 Hz	20 ms	160 B (μ-law silence 0xFF)
PCMA/8000	8	8 000 Hz	20 ms	160 B (A-law silence 0xD5)
G722/8000	9	8 000 Hz	20 ms	160 B (zero)
ILBC/8000	97	8 000 Hz	30 ms	240 B (zero)
H264/90000	96	90 000 Hz	33 ms	3000 B (zero)
OPUS/48000	111	48 000 Hz	20 ms	3 B (Opus DTX silence frame)

Dynamic payload types — for ILBC, H264 and Opus the PT values above are common defaults. Use -rtp_pt to override when the remote party negotiates a different PT in the SDP offer.

Examples

# Send PCMU silence for 5 seconds at 20 ms intervals
gossipper sipp -rtp_send -rtp_addr 192.168.1.10:20000 -rtp_dur 5000

# Send PCMA at 20 ms intervals until Ctrl-C
gossipper sipp -rtp_send -rtp_addr 192.168.1.10:20000 -rtp_codec PCMA/8000

# G722 with a non-default dynamic PT, 10 seconds
gossipper sipp -rtp_send -rtp_addr 192.168.1.10:20000 \
  -rtp_codec G722/8000 -rtp_pt 110 -rtp_dur 10000

# 30 ms ILBC frames, stereo channel layout, 1 minute
gossipper sipp -rtp_send -rtp_addr 192.168.1.10:20000 \
  -rtp_codec ILBC/8000 -rtp_ch 2 -rtp_dur 60000

# Bind to a specific local IP
gossipper sipp -rtp_send -rtp_addr 192.168.1.10:20000 -i 10.0.0.5 -rtp_dur 3000

Output

Gossipper prints a start line and a summary when the stream stops:

rtp_sender: streaming to 192.168.1.10:20000  codec=PCMA/8000 pt=8 freq=20ms duration=5000ms channels=1
rtp_sender: done  sent=250 octets=40000 rtcp_sr=0

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Scenario action: rtp_stream="synthetic,…"

The synthetic keyword can be used anywhere rtp_stream is accepted inside an XML scenario. It follows the same comma-separated parameter convention as file-based streaming but requires no file path.

Syntax

rtp_stream="synthetic[,<loop_count>[,<pt>[,<codec_name>[,<freq_ms>[,<dur_ms>[,<channels>]]]]]]"

All parameters after synthetic are optional and fall back to the same defaults as the standalone CLI mode.

Parameters

Position	Name	Default	Description
1	loop_count	—	Accepted for spec symmetry but ignored; dur_ms (or context cancellation) controls the stop
2	pt	from codec	RTP payload type number
3	codec_name	PCMU/8000	Codec descriptor (same table as CLI)
4	freq_ms	20	Packet interval in milliseconds
5	dur_ms	0	Duration in milliseconds; 0 = stream until stop action or call teardown
6	channels	1	Number of audio channels

Examples

<!-- PCMU silence, default 20 ms, run until stop -->
<exec rtp_stream="synthetic"/>

<!-- PCMA, 20 ms intervals, stop after 5 seconds -->
<exec rtp_stream="synthetic,0,8,PCMA/8000,20,5000,1"/>

<!-- G722, 20 ms intervals, 10 seconds, stereo -->
<exec rtp_stream="synthetic,0,9,G722/8000,20,10000,2"/>

<!-- ILBC, 30 ms intervals, run until stop -->
<exec rtp_stream="synthetic,0,97,ILBC/8000,30,0,1"/>

<!-- H264, 33 ms intervals, 30 seconds -->
<exec rtp_stream="synthetic,0,96,H264/90000,33,30000,1"/>

Control commands (pause, resume, stop, echo) work the same way as with file-based streams:

<exec rtp_stream="pause"/>
<exec rtp_stream="resume"/>
<exec rtp_stream="stop"/>

Complete scenario example

The following UAC scenario sends synthetic PCMU media for the duration of the call without requiring any audio file on disk:

<?xml version="1.0" encoding="UTF-8" ?>
<scenario name="synthetic-audio-uac">

  <send retrans="500">
    <![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len]

v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
    ]]>
  </send>

  <recv response="100" optional="true"/>

  <recv response="200" rtd="true">
    <action>
      <!-- Start synthetic PCMU stream; no audio file needed -->
      <exec rtp_stream="synthetic,0,0,PCMU/8000,20,0,1"/>
    </action>
  </recv>

  <send>
    <![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: [cseq] ACK
Max-Forwards: 70
Content-Length: 0
    ]]>
  </send>

  <pause milliseconds="4000"/>

  <nop>
    <action>
      <exec rtp_stream="stop"/>
    </action>
  </nop>

  <send retrans="500">
    <![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: [cseq] BYE
Max-Forwards: 70
Content-Length: 0
    ]]>
  </send>

  <recv response="200"/>

</scenario>

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Payload content

Synthetic payloads contain codec-appropriate silence bytes:

Codec	Payload	Rationale
PCMU (G.711 μ-law)	0xFF × SamplesPerPkt × Channels	μ-law encoding of zero-amplitude signal
PCMA (G.711 A-law)	0xD5 × SamplesPerPkt × Channels	A-law encoding of zero-amplitude signal
Opus	{0xF8, 0xFF, 0xFE} (3 bytes)	Valid CELT FB 20 ms mono DTX frame (RFC 6716)
All others	0x00 × SamplesPerPkt × Channels	Zero-filled; most codecs treat this as silence or a degenerate frame

For PCM-based codecs (PCMU, PCMA, G722) the payload size is SamplesPerPkt × Channels. For mono PCMU/PCMA at 20 ms this is 160 × 1 = 160 bytes per packet.

For Opus, the payload is always 3 bytes regardless of channel count. Opus is a variable-bitrate codec; its frames are not raw PCM and their size does not follow the PCM formula. The SamplesPerPkt value (960 for 20 ms at 48 kHz) is used only for the RTP timestamp increment.

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Channels

The -rtp_ch flag (CLI) or the channels parameter (scenario) multiplies the per-packet payload size:

payload_bytes = samples_per_packet × channels

RTP itself is not inherently channel-aware; the channel count here simply controls the byte count placed in each packet payload. The remote endpoint must be configured to expect the same frame size.

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Interaction with HEP mirroring

When -hep_addr is configured and -send_media_report is enabled, synthetic RTP packets are mirrored to the HEP collector the same way file-based packets are. No additional configuration is needed.

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Differences from file-based rtp_stream

Aspect	File-based	Synthetic
Requires a media file	Yes (.wav or .raw)	No
Payload content	Actual audio samples	Codec-appropriate silence
Loop control	loop_count; -1 = infinite	Ignored; use dur_ms or stop
Duration control	Derived from file length × loop count	dur_ms parameter or context cancellation
Codec parameters	Parsed from file format + spec	Fully configurable via spec

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Known limits

• Synthetic streams always produce silence; arbitrary tone or noise generation is not supported.
• SRTP / DTLS: synthetic rtp_stream uses the same media session as file-based audio; enable -media_srtp when the peer SDP requires SDES or DTLS-SRTP (see srtp.md).
• Multi-channel WAV encoding is not validated against the channels parameter; the parameter only affects the raw byte count of each synthetic payload.