"The Codenomicon tools are amazing. Using them is like being attacked by the most relentless adversary who uses every possible method to find flaws in your code
We fixed subtle crash bugs in Samba that had been in the code for over ten years. We would never have found those bugs without the Codenomicon tools.
If you're serious about implementing protocols correctly, you need the Codenomicon tools."
-- Jeremy Allison,
Co Creator of Samba
Products by Protocol
Codenomicon RTP Test Tool Data Sheet
- Test tool: Codenomicon RTP Test Tool
- Direction: Server
- Tagline: Improving the Robustness of Real-Time Traffic
With the Codenomicon RTP Test Tool, you can proactively eliminate robustness flaws from your critical VoIP and IPTV infrastructure. This saves development and maintenance costs, produces more stable systems, and enhances your product and corporate image. The Codenomicon RTP Test Tool is essential for anyone who develops VoIP or IPTV applications, or depends on the robust functioning of real-time transport services.
RTP (Real-time Transport Protocol) is a protocol for the transmission of real-time data over the Internet. Examples of the data carried by RTP typically include audio, video and other streaming multimedia content. RTP is augmented by RTCP (RTP Control Protocol), which provides control, identification and monitoring for RTP data delivery.
Used specifications
| Specification | Title |
|---|---|
| RFC2326 | Real Time Streaming Protocol (RTSP) |
| RFC2327 | SDP: Session Description Protoco |
| RFC2617 | HTTP Authentication: Basic and Digest Access Authentication |
| RFC2833 | RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals |
| RFC3261 | SIP: Session Initiation Protocol |
| RFC3389 | Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN) |
| RFC3435 | Media Gateway Control Protocol (MGCP) Version 1.0 |
| RFC3550 | RTP: A Transport Protocol for Real-Time Applications |
| RFC3551 | RTP Profile for Audio and Video Conferences with Minimal Control |
| RFC3611 | RTP Control Protocol Extended Reports (RTCP XR) |
| RFC3640 | RTP Payload Format for Transport of MPEG-4 Elementary Streams |
| RFC3711 | The Secure Real-time Transport Protocol (SRTP) |
| RFC3830 | MIKEY: Multimedia Internet KEYing |
| RFC4567 | Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol (RTSP) |
| RFC4568 | Session Description Protocol Security Descriptions for Media Streams |
| RFC4733 | RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals |
| OMA-UP-V1_0 | OMA PoC User Plane Signaling (OMA-TS-PoC-UserPlane-V1_0-20060609-A) |
| 3GPP-TS-26.234-V6.7.0 | Technical Specification 3rd Generation Partnership Project; Technical Specification Group Services and System Aspects; Transparent end-to-end Packet-switched Streaming Service (PSS); Protocols and codecs (Release 6) |
Test tool general features
- Fully automated black-box negative testing
- Ready-made test cases
- Written in Java(tm)
- GUI, command line, remote interface modes
- Instrumentation (health-check) capability
- Support and maintenance
- Comprehensive user documentation
- Results reporting and analysis
Tool-specific information
| Tested messages | Notes | Specifications |
|---|---|---|
| RTP-Data-Transfer-Protocol | Framing used for transfering RTP payload formats | RFC3550 |
| SRTP | Secure RTP | RFC3771 |
| RTP-Control-Protocol/SR | Sender Report | RFC3550 |
| RTP-Control-Protocol/RR | Receiver Report | RFC3550 |
| RTP-Control-Protocol/SDES | Source Description | RFC3550 |
| RTP-Control-Protocol/BYE | Bye | RFC3550 |
| RTP-Control-Protocol/APP | Application (generic application framing) | RFC3550 |
| RTP-Control-Protocol/XR | Extended Report | RFC3611 |
| SRTCP | Secure RTCP | RFC3771 |
| RTCP/APP/TBCP-Connect | Talk Burst Control Protocol: Connect | OMA-UP-V1_0 |
| RTCP/APP/TBCP-Disconnect | Talk Burst Control Protocol: Disconnect | OMA-UP-V1_0 |
| RTCP/APP/TBCP-TB-Request | Talk Burst Control Protocol: Talk Burst Request | OMA-UP-V1_0 |
| RTCP/APP/TBCP-TB-Granted | Talk Burst Control Protocol: Talk Burst Granted | OMA-UP-V1_0 |
| RTCP/APP/TBCP-TB-Deny | Talk Burst Control Protocol: Talk Burst Deny | OMA-UP-V1_0 |
| RTCP/APP/TBCP-TB-Release | Talk Burst Control Protocol: Talk Burst Release | OMA-UP-V1_0 |
| RTCP/APP/TBCP-TB-Idle | Talk Burst Control Protocol: Talk Burst Idle | OMA-UP-V1_0 |
| RTCP/APP/TBCP-TB-Taken | Talk Burst Control Protocol: Talk Burst Taken | OMA-UP-V1_0 |
| RTCP/APP/TBCP-TB-Revoke | Talk Burst Control Protocol: Talk Burst Revoke | OMA-UP-V1_0 |
| RTCP/APP/TBCP-TB-Ack | Talk Burst Control Protocol: Talk Burst Acknowledgement | OMA-UP-V1_0 |
| RTCP/APP/TBCP-TB-Queue-Req | Talk Burst Control Protocol: Talk Burst Queue Status Request | OMA-UP-V1_0 |
| RTCP/APP/TBCP-TB-Queue-Res | Talk Burst Control Protocol: Talk Burst Queue Status Response | OMA-UP-V1_0 | Supported RTP initiators | Notes | Specifications |
| Pure RTP | RTP/RTCP datagrams are sent directly to a defined target | RFC3550 |
| Session Initiation Protocol (SIP) signaling | Setup RTP channel using SIP and SDP | RFC3261 |
| Real Time Streaming Protocol (RTSP) signaling | Setup RTP channel using RTSP | RFC2326 |
| Media Gateway Control Protocol (MGCP) signaling | Setup RTP channel using MGCP and SDP | RFC3435 | Supported RTP payloads | Notes | Available since |
| G711-PCMU | 1.0 | |
| G711-PCMA | 2.1 | |
| G721 / G726-32 | 2.1 | |
| G729 | 2.1 | |
| GSM 06.10 | 1.0.2 | |
| iLBC | 1.0.2 | |
| AMR | 1.0.2 | |
| MPEG4-audio | ISO/IEC 14496-3 | 2.0 |
| RFC4733 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals | Obsoletes RFC2833 | 2.1 |
| RFC3389 Comfort Noise | 2.1 | |
| MPEG4-video | ISO/IEC 14496-2 | 2.3 |
| H.264 | 2.3 |
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