RTP - Real-Time Transport Protocol
What is RTP?
RTP, Real-Time Transport Protocol, is a network protocol standardized by the IETF (RFC 3550) for delivering audio and video over IP networks. It is widely used in streaming media systems, video conferencing, and VoIP (Voice over IP).
Why RTP is Useful?
Supports real-time transmission of multimedia data.
Works with RTCP for quality monitoring and synchronization.
Payload type identification allows dynamic codec switching.
Timestamping and sequence numbers enable jitter compensation and packet reordering.
Widely used in VoIP, WebRTC, and conferencing systems.
How it works?
Media is encoded – Audio/video is compressed using codecs (e.g., Opus, H.264).
Packets are created – RTP adds headers with sequence numbers and timestamps.
Data is transmitted – Packets are sent over UDP (usually) to minimize latency.
Receiver reorders and plays – The client uses timestamps and sequence numbers to reconstruct the stream.
RTCP feedback – RTCP provides stats like packet loss, jitter, and round-trip time.
Where is RTP used?
VoIP systems – SIP-based voice calls (e.g., Skype, Zoom, WhatsApp).
Video conferencing – Google Meet, Microsoft Teams, WebRTC-based apps.
Streaming media – Used in conjunction with RTSP for IP camera feeds.
Which OSI layer does this protocol belong to?
RTP is considered an application-layer protocol because it provides end-to-end delivery services for real-time data.
It relies on lower-layer protocols like UDP for transport.
RTP handles timing, sequencing, and payload identification, which are application-level concerns.
Topics in this section,
In this section, you are going to learn
Terminology
Version Info
sno |
RTP Version |
RFC / Standard |
Core / Contribution |
Year |
---|---|---|---|---|
1 |
RTP v1 |
RFC 1889 RTP: A Transport Protocol for Real-Time Applications |
|
1996 |
2 |
RTP v2 (Current) |
RFC 3550 Obsoletes RFC 1889 |
Same as above |
2003 |
3 |
RTP Control Protocol (RTCP) |
RFC 3550 (Part of RTP v2) |
Same as above |
2003 |
4 |
Secure RTP (SRTP) |
RFC 3711 |
|
2004 |
5 |
RTP Retransmission |
RFC 4588 |
|
2006 |
6 |
RTP over TCP |
RFC 4571 |
|
2006 |
7 |
RTP Payload for MPEG-4 |
RFC 3016 |
|
2001 |
8 |
RTP Payload for H.264 Video |
RFC 6184 |
|
2011 |
9 |
RTP Payload for VP8 Video |
RFC 7741 |
|
2016 |
10 |
RTP Payload for VP9 Video |
RFC 8627 |
|
2019 |
11 |
RTP Payload for AV1 Video |
RFC 9328 |
|
2022 |
12 |
RTP Payload for Opus Audio |
RFC 7587 |
|
2015 |
13 |
RTP Header Extension for Audio Level |
RFC 6464 |
|
2011 |
14 |
RTP Header Extension for Absolute Capture Time |
RFC 8872 |
|
2021 |
15 |
RTP with RTCP Multiplexing |
RFC 5761 |
|
2010 |
16 |
RTP Redundant Audio Data |
RFC 2198 |
|
1997 |
17 |
RTP Stream Identifier Source Description |
RFC 8852 |
|
2020 |
Setup
Setup
RTP Header Packet
S.No |
Protocol Packets |
Description |
Size(Bytes) |
---|---|---|---|
1 |
RTP Header |
Contains control information for the RTP stream |
12 bytes |
Version (V) |
RTP version (usually 2) |
2 bits |
|
Padding (P) |
Indicates if padding bytes are added at the end |
1 bit |
|
Extension (X) |
Indicates presence of extension header |
1 bit |
|
CSRC Count (CC) |
Number of CSRC identifiers |
4 bits |
|
Marker (M) |
Application-specific marker bit |
1 bit |
|
Payload Type (PT) |
Identifies the format of the payload (e.g., codec) |
7 bits |
|
Sequence Number |
Increments by one for each RTP packet sent |
2 bytes |
|
Timestamp |
Sampling instant of the first byte in the payload |
4 bytes |
|
SSRC Identifier |
Identifies the synchronization source |
4 bytes |
|
CSRC Identifiers |
List of contributing sources (if any) |
015 entries, 4 bytes each |
CSRC List Packet
S.No |
Protocol Packets |
Description |
Size(Bytes) |
---|---|---|---|
2 |
CSRC List (optional) |
Identifies contributing sources in a conference |
0 to 60 bytes |
CSRC Identifier(s) |
Identifies contributing sources in a conference |
4 bytes each (up to 15 entries) |
Extension Header Packet
S.No |
Protocol Packets |
Description |
Size(Bytes) |
---|---|---|---|
3 |
Extension Header (optional) |
Additional application-specific data |
Variable |
Extension Header ID |
Identifies the extension format |
2 bytes |
|
Extension Length |
Number of 32-bit words in the extension |
2 bytes |
|
Extension Data |
Application-specific data |
Variable |
Payload Packet
S.No |
Protocol Packets |
Description |
Size(Bytes) |
---|---|---|---|
Payload (Media Data) |
Actual audio/video data |
Variable |
|
4 |
Encoded Media Data |
Actual audio or video content (e.g., G.711, H.264) |
Variable (depends on codec and MTU) |
Padding Packet
S.No |
Protocol Packets |
Description |
Size(Bytes) |
---|---|---|---|
5 |
Padding (optional) |
Extra bytes for alignment or encryption |
Variable |
Padding Bytes |
Extra bytes added for alignment or encryption |
Variable |
|
Last Byte |
Indicates the number of padding bytes |
1 byte |
sno |
Use Case |
Description |
---|---|---|
1 |
VoIP (Voice over IP) |
RTP is the backbone of voice communication over IP networks, used in SIP-based systems like Skype, Zoom, and WhatsApp. |
2 |
Video Conferencing |
Enables real-time video and audio transmission in platforms like Microsoft Teams, Google Meet, and WebRTC-based apps. |
3 |
Live Streaming |
Used in conjunction with RTSP or RTMP for low-latency live video delivery in surveillance, events, and broadcasting. |
4 |
IPTV (Internet Protocol Television) |
Delivers live TV and video-on-demand services over IP networks using RTP for media transport. |
5 |
Surveillance Systems |
IP cameras use RTP to stream real-time video feeds to monitoring systems or cloud storage. |
6 |
Telemedicine |
Facilitates real-time audio/video communication between doctors and patients in remote healthcare setups. |
7 |
Online Education |
Powers live virtual classrooms and webinars with real-time interaction between instructors and students. |
8 |
Gaming Voice Chat |
Used in multiplayer games for real-time voice communication between players. |
9 |
Broadcast Contribution Feeds |
Used to send high-quality, low-latency audio/video from field reporters or remote studios to central broadcast centers. |
10 |
Remote Desktop and Virtualization |
Supports real-time audio/video in remote desktop protocols like Citrix and VMware Horizon. |
11 |
Military and Tactical Communications |
Used in secure, real-time communication systems for defense and emergency response. |
12 |
Smart Home and IoT Devices |
RTP is used in smart doorbells, intercoms, and home surveillance for real-time audio/video streaming. |
13 |
Augmented and Virtual Reality (AR/VR) |
Supports real-time media delivery in immersive environments requiring low latency. |
14 |
Interactive Kiosks and Digital Signage |
Enables real-time content updates and live feeds in public displays and kiosks. |
15 |
Media Gateways and Transcoders |
RTP is used to transport media between different network types and codecs in telecom infrastructure. |
sno |
Feature |
Description |
---|---|---|
1 |
Real-Time Delivery |
Designed for transmitting audio and video with minimal delay over IP networks. |
2 |
Timestamping |
Each packet includes a timestamp to help synchronize playback and manage jitter. |
3 |
Sequence Numbering |
Packets are numbered to detect loss and reorder out-of-sequence packets. |
4 |
Payload Type Identification |
Identifies the codec used (e.g., H.264, Opus), allowing dynamic media format handling. |
5 |
Source Identification (SSRC) |
Each stream has a unique identifier to distinguish multiple sources. |
6 |
Contributing Source List (CSRC) |
Supports mixing multiple sources in conferencing scenarios. |
7 |
Header Extensions |
Allows custom metadata to be added for advanced features like audio levels or capture time. |
8 |
Support for Multicast and Unicast |
Can be used in both point-to-point and group communication setups. |
9 |
Works with RTCP |
RTCP provides feedback on quality, synchronization, and participant information. |
10 |
Codec Agnostic |
RTP does not enforce any specific codec, allowing flexibility in media encoding. |
11 |
Transport Flexibility |
Typically runs over UDP but can also be used over TCP or other transports. |
12 |
Extensibility |
Easily extended with new payload formats and header extensions. |
13 |
Interoperability |
Widely supported across platforms, devices, and applications. |
14 |
Security via SRTP |
Can be secured using Secure RTP (SRTP) for encryption and authentication. |
15 |
Jitter Compensation |
Timestamp and sequence numbers help receivers manage network jitter. |
Real-Time Delivery - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
RTP Packet Transmission |
Send RTP packets |
Packets are delivered in real-time |
2 |
RTP Timestamp Accuracy |
Verify timestamps |
Timestamps match media clock |
3 |
RTP Sequence Numbering |
Check sequence numbers |
Packets are ordered correctly |
4 |
RTP Clock Synchronization |
Sync sender and receiver clocks |
Synchronization is maintained |
5 |
RTP with RTCP Feedback |
Use RTCP for control |
Feedback is received timely |
6 |
RTP with Jitter Buffer |
Use jitter buffer at receiver |
Playback is smooth |
7 |
RTP with Packet Loss |
Simulate packet loss |
RTP handles loss gracefully |
8 |
RTP with Packet Reordering |
Reorder packets |
RTP reorders correctly |
9 |
RTP with Delay Variation |
Introduce delay variation |
RTP adapts to delay |
10 |
RTP with High Latency |
Simulate 500ms latency |
Stream continues with delay |
11 |
RTP with Low Latency |
Simulate <50ms latency |
Stream plays in near real-time |
12 |
RTP with Network Congestion |
Simulate congestion |
RTP adapts or degrades gracefully |
13 |
RTP with Bandwidth Throttling |
Limit bandwidth |
RTP adjusts bitrate |
14 |
RTP with Encryption |
Use SRTP |
Packets are encrypted and delivered |
15 |
RTP with Authentication |
Authenticate session |
Only authorized packets accepted |
16 |
RTP with NAT Traversal |
Use STUN/TURN |
RTP packets traverse NAT |
17 |
RTP with Firewall |
Send through firewall |
Packets are delivered |
18 |
RTP with IPv4 |
Use IPv4 network |
RTP packets are delivered |
19 |
RTP with IPv6 |
Use IPv6 network |
RTP packets are delivered |
20 |
RTP with UDP |
Use UDP transport |
Real-time delivery is maintained |
21 |
RTP with TCP |
Use TCP transport |
Delivery is slower but reliable |
22 |
RTP with RTP-MUX |
Use multiplexed RTP |
Multiple streams delivered correctly |
23 |
RTP with RTSP |
Use RTSP for control |
RTP stream is managed |
24 |
RTP with SIP |
Use SIP for session setup |
RTP stream is established |
25 |
RTP with WebRTC |
Use RTP in WebRTC |
Real-time delivery is achieved |
26 |
RTP with VoIP |
Use RTP for voice call |
Audio is delivered in real-time |
27 |
RTP with Video Streaming |
Use RTP for video |
Video plays smoothly |
28 |
RTP with Audio Streaming |
Use RTP for audio |
Audio plays smoothly |
29 |
RTP with Multi-Channel Audio |
Stream 5.1 audio |
Channels are synchronized |
30 |
RTP with Multi-Resolution Video |
Stream adaptive video |
Resolution switches smoothly |
31 |
RTP with Subtitles |
Send subtitles via RTP |
Subtitles are synchronized |
32 |
RTP with Metadata |
Send metadata via RTP |
Metadata is received in real-time |
33 |
RTP with Live Broadcast |
Stream live event |
Minimal delay in delivery |
34 |
RTP with On-Demand Playback |
Play recorded RTP stream |
Playback is accurate |
35 |
RTP with Session Timeout |
Let session expire |
RTP stream ends gracefully |
36 |
RTP with Session Reconnect |
Reconnect session |
RTP resumes delivery |
37 |
RTP with Packet Duplication |
Duplicate packets |
RTP discards duplicates |
38 |
RTP with Packet Corruption |
Corrupt packets |
RTP detects and handles errors |
39 |
RTP with QoS Enabled |
Enable QoS |
RTP packets prioritized |
40 |
RTP with DSCP Marking |
Use DSCP for priority |
Packets marked correctly |
41 |
RTP with Multicast |
Use multicast delivery |
Packets received by all clients |
42 |
RTP with Unicast |
Use unicast delivery |
Packets received by single client |
43 |
RTP with Load Balancer |
Route through load balancer |
Delivery remains real-time |
44 |
RTP with CDN |
Use CDN for delivery |
Packets delivered from edge nodes |
45 |
RTP with Mobile Network |
Use 4G/5G |
RTP adapts to mobile conditions |
46 |
RTP with Wi-Fi |
Use wireless network |
RTP maintains delivery |
47 |
RTP with Ethernet |
Use wired network |
RTP delivers with minimal delay |
48 |
RTP with VPN |
Use VPN tunnel |
RTP packets delivered securely |
49 |
RTP with Proxy |
Use HTTP proxy |
RTP packets are tunneled |
50 |
RTP with Monitoring Tools |
Use RTP monitoring |
Delivery metrics are logged |
Timestamping - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
RTP Timestamp Initialization |
Check first packet timestamp |
Timestamp starts from a valid base |
2 |
RTP Timestamp Increment |
Verify timestamp increases |
Increments match media clock rate |
3 |
RTP Timestamp Wraparound |
Simulate 32-bit wraparound |
Receiver handles wrap correctly |
4 |
RTP Timestamp with Audio |
Send audio stream |
Timestamps align with sample rate |
5 |
RTP Timestamp with Video |
Send video stream |
Timestamps align with frame rate |
6 |
RTP Timestamp with Silence |
Send silent audio |
Timestamps continue incrementing |
7 |
RTP Timestamp with Frame Drop |
Drop video frames |
Timestamps remain consistent |
8 |
RTP Timestamp with Packet Loss |
Simulate loss |
Receiver interpolates timestamps |
9 |
RTP Timestamp with Delay |
Introduce network delay |
Timestamps remain unaffected |
10 |
RTP Timestamp with Jitter |
Simulate jitter |
Jitter buffer uses timestamps |
11 |
RTP Timestamp with RTCP SR |
Compare with RTCP Sender Report |
Timestamps match NTP time |
12 |
RTP Timestamp with RTCP RR |
Use Receiver Report |
Delay and jitter are calculated |
13 |
RTP Timestamp with Playback Sync |
Sync audio and video |
Streams are synchronized |
14 |
RTP Timestamp with Clock Drift |
Simulate drift |
Receiver compensates |
15 |
RTP Timestamp with Pause/Resume |
Pause and resume stream |
Timestamps continue correctly |
16 |
RTP Timestamp with Seek |
Seek in media |
New timestamps are valid |
17 |
RTP Timestamp with Rewind |
Rewind stream |
Timestamps reset appropriately |
18 |
RTP Timestamp with Fast Forward |
Skip ahead |
Timestamps jump forward |
19 |
RTP Timestamp with Multicast |
Send to multiple clients |
All receive consistent timestamps |
20 |
RTP Timestamp with Unicast |
Send to single client |
Timestamps are accurate |
21 |
RTP Timestamp with Encryption |
Use SRTP |
Timestamps are preserved |
22 |
RTP Timestamp with Compression |
Compress stream |
Timestamps remain accurate |
23 |
RTP Timestamp with Variable Bitrate |
Change bitrate |
Timestamps remain consistent |
24 |
RTP Timestamp with Constant Bitrate |
Use CBR stream |
Timestamps increment uniformly |
25 |
RTP Timestamp with High Frame Rate |
Use 60fps video |
Timestamps reflect frame rate |
26 |
RTP Timestamp with Low Frame Rate |
Use 15fps video |
Timestamps reflect frame rate |
27 |
RTP Timestamp with Audio Sample Rate |
Use 44.1kHz audio |
Timestamps increment by 441 samples |
28 |
RTP Timestamp with 48kHz Audio |
Use 48kHz audio |
Timestamps increment by 480 samples |
29 |
RTP Timestamp with Silence Suppression |
Suppress silent packets |
Timestamps skip accordingly |
30 |
RTP Timestamp with Packet Duplication |
Duplicate packets |
Receiver ignores duplicates |
31 |
RTP Timestamp with Out-of-Order Packets |
Reorder packets |
Playback uses correct timestamps |
32 |
RTP Timestamp with RTSP |
Use RTSP control |
Timestamps align with playback |
33 |
RTP Timestamp with SIP |
Use SIP session |
Timestamps are synchronized |
34 |
RTP Timestamp with WebRTC |
Use WebRTC |
Timestamps support real-time sync |
35 |
RTP Timestamp with Live Streaming |
Stream live event |
Timestamps reflect real-time |
36 |
RTP Timestamp with VOD |
Play recorded stream |
Timestamps match media timeline |
37 |
RTP Timestamp with Transcoding |
Transcode stream |
Timestamps are recalculated |
38 |
RTP Timestamp with Adaptive Streaming |
Change quality |
Timestamps remain continuous |
39 |
RTP Timestamp with Multi-Track Audio |
Use multiple audio tracks |
Timestamps are track-specific |
40 |
RTP Timestamp with Multi-Camera Video |
Switch camera feeds |
Timestamps remain in sync |
41 |
RTP Timestamp with Subtitles |
Send timed captions |
Captions align with media |
42 |
RTP Timestamp with Metadata |
Send timed metadata |
Metadata aligns with media |
43 |
RTP Timestamp with Monitoring Tools |
Use RTP analyzer |
Timestamps are logged correctly |
44 |
RTP Timestamp with Logging |
Log timestamps |
Logs show correct values |
45 |
RTP Timestamp with Debug Mode |
Enable debug |
Timestamps are printed |
46 |
RTP Timestamp with Session Timeout |
Let session expire |
Final timestamp is valid |
47 |
RTP Timestamp with Reconnect |
Reconnect session |
Timestamps resume correctly |
48 |
RTP Timestamp with Clock Reset |
Reset sender clock |
Receiver handles jump |
49 |
RTP Timestamp with NTP Sync |
Sync with NTP |
Timestamps align with wall clock |
50 |
RTP Timestamp with Custom Clock Rate |
Use non-standard rate |
Timestamps increment accordingly |
Sequence Numbering - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
Valid Sequence Start |
RTP packet starts with sequence number 0 |
Sequence number is 0 |
2 |
Increment Sequence |
Next RTP packet increments sequence number by 1 |
Sequence number is previous + 1 |
3 |
Wrap Around |
Sequence number wraps after 65535 |
Sequence number resets to 0 |
4 |
Duplicate Packet |
Same sequence number received twice |
Packet marked as duplicate |
5 |
Out-of-Order Packet |
Sequence number received out of order |
Packet marked as out-of-order |
6 |
Missing Packet |
Sequence number skipped |
Packet loss detected |
7 |
Sequence Reset |
RTP stream reset, sequence starts from 0 |
Sequence number is 0 |
8 |
Large Jump |
Sequence number jumps by more than 100 |
Possible stream corruption |
9 |
Negative Sequence |
Sequence number is negative |
Invalid packet |
10 |
Non-integer Sequence |
Sequence number is float |
Invalid packet |
11 |
Sequence in Header |
Sequence number present in RTP header |
Header parsed correctly |
12 |
Sequence with Jitter |
Sequence number affected by jitter |
Jitter compensation applied |
13 |
Sequence with Delay |
Delayed packet with correct sequence |
Packet accepted |
14 |
Sequence with Timestamp |
Sequence number matches timestamp order |
Synchronization verified |
15 |
Sequence with SSRC Change |
SSRC changes mid-stream |
Sequence reset expected |
16 |
Sequence with Payload Change |
Payload type changes |
Sequence continuity maintained |
17 |
Sequence with Marker Bit |
Marker bit set on key frame |
Sequence number verified |
18 |
Sequence with Encryption |
Encrypted RTP packet |
Sequence number decrypted correctly |
19 |
Sequence with Extension Header |
RTP extension used |
Sequence number unaffected |
20 |
Sequence with RTCP Feedback |
RTCP reports packet loss |
Sequence number used for feedback |
21 |
Sequence with Retransmission |
Retransmitted packet has same sequence |
Packet marked as retransmission |
22 |
Sequence with FEC |
Forward Error Correction used |
Sequence number used for recovery |
23 |
Sequence with SRTP |
Secure RTP used |
Sequence number integrity verified |
24 |
Sequence with Multicast |
RTP over multicast |
Sequence number synchronized across receivers |
25 |
Sequence with Unicast |
RTP over unicast |
Sequence number continuity maintained |
26 |
Sequence with NAT |
RTP through NAT |
Sequence number unaffected |
27 |
Sequence with Firewall |
RTP through firewall |
Sequence number continuity maintained |
28 |
Sequence with VPN |
RTP over VPN |
Sequence number integrity verified |
29 |
Sequence with IPv6 |
RTP over IPv6 |
Sequence number parsed correctly |
30 |
Sequence with IPv4 |
RTP over IPv4 |
Sequence number parsed correctly |
31 |
Sequence with QoS |
Quality of Service enabled |
Sequence number used for prioritization |
32 |
Sequence with Congestion |
Network congestion |
Sequence number gaps detected |
33 |
Sequence with Packet Duplication |
Network duplicates packets |
Duplicate sequence detected |
34 |
Sequence with Packet Reordering |
Network reorders packets |
Out-of-order sequence detected |
35 |
Sequence with Packet Loss |
Network drops packets |
Missing sequence detected |
36 |
Sequence with Buffer Overflow |
Receiver buffer overflow |
Sequence number mismatch |
37 |
Sequence with Buffer Underflow |
Receiver buffer underflow |
Sequence number gaps |
38 |
Sequence with Clock Drift |
Sender clock drift |
Sequence number and timestamp mismatch |
39 |
Sequence with Synchronization |
Multiple streams synchronized |
Sequence numbers aligned |
40 |
Sequence with RTP Mixer |
RTP mixer modifies stream |
Sequence number continuity maintained |
41 |
Sequence with Translator |
RTP translator modifies stream |
Sequence number continuity maintained |
42 |
Sequence with Silence Suppression |
Silence periods skipped |
Sequence number continuity maintained |
43 |
Sequence with DTMF Events |
DTMF tones sent via RTP |
Sequence number continuity maintained |
44 |
Sequence with Video Stream |
RTP used for video |
Sequence number used for frame ordering |
45 |
Sequence with Audio Stream |
RTP used for audio |
Sequence number used for sample ordering |
46 |
Sequence with High Bitrate |
High bitrate stream |
Sequence number increments rapidly |
47 |
Sequence with Low Bitrate |
Low bitrate stream |
Sequence number increments slowly |
48 |
Sequence with Burst Traffic |
RTP packets sent in bursts |
Sequence number continuity verified |
49 |
Sequence with Real-Time Monitoring |
RTP monitored in real-time |
Sequence number used for diagnostics |
50 |
Sequence with Logging |
RTP sequence numbers logged |
Sequence number recorded accurately |
Payload Type Identification - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
Valid Payload Type |
RTP packet with payload type 0 (PCMU) |
Payload identified as PCMU |
2 |
Dynamic Payload Type |
Payload type 96 used for dynamic codec |
Payload mapped via signaling |
3 |
Invalid Payload Type |
Payload type 200 used |
Packet rejected or flagged |
4 |
Audio Payload |
Payload type 8 (PCMA) |
Identified as PCMA audio |
5 |
Video Payload |
Payload type 34 (H.263) |
Identified as H.263 video |
6 |
Payload Type Change |
Payload type changes mid-stream |
Change detected and handled |
7 |
Payload Type with SSRC Change |
SSRC changes with same payload |
Payload type still identified |
8 |
Payload Type with Marker Bit |
Marker bit set on key frame |
Payload type unaffected |
9 |
Payload Type with Extension Header |
RTP extension present |
Payload type correctly parsed |
10 |
Payload Type with Encryption |
Encrypted RTP packet |
Payload type decrypted and identified |
11 |
Payload Type with SRTP |
Secure RTP used |
Payload type integrity verified |
12 |
Payload Type with RTCP Feedback |
RTCP reports payload type |
Payload type confirmed |
13 |
Payload Type with Silence Suppression |
Silence periods skipped |
Payload type continuity maintained |
14 |
Payload Type with DTMF |
Payload type 101 (DTMF) |
Identified as DTMF event |
15 |
Payload Type with Comfort Noise |
Payload type 13 (CN) |
Identified as comfort noise |
16 |
Payload Type with Redundancy |
Payload type 122 (RED) |
Identified as redundant encoding |
17 |
Payload Type with FEC |
Payload type 123 (FEC) |
Identified as FEC |
18 |
Payload Type with Jitter |
Jitter affects packet timing |
Payload type still identified |
19 |
Payload Type with Delay |
Delayed packet |
Payload type correctly parsed |
20 |
Payload Type with Packet Loss |
Some packets lost |
Payload type identified from received packets |
21 |
Payload Type with Reordering |
Packets arrive out of order |
Payload type correctly identified |
22 |
Payload Type with Duplication |
Duplicate packets |
Payload type remains consistent |
23 |
Payload Type with Multicast |
RTP over multicast |
Payload type identified across receivers |
24 |
Payload Type with Unicast |
RTP over unicast |
Payload type correctly parsed |
25 |
Payload Type with NAT |
RTP through NAT |
Payload type unaffected |
26 |
Payload Type with Firewall |
RTP through firewall |
Payload type correctly parsed |
27 |
Payload Type with VPN |
RTP over VPN |
Payload type integrity maintained |
28 |
Payload Type with IPv6 |
RTP over IPv6 |
Payload type correctly parsed |
29 |
Payload Type with IPv4 |
RTP over IPv4 |
Payload type correctly parsed |
30 |
Payload Type with QoS |
QoS enabled |
Payload type used for prioritization |
31 |
Payload Type with Congestion |
Network congestion |
Payload type still identified |
32 |
Payload Type with Translator |
RTP translator modifies stream |
Payload type updated accordingly |
33 |
Payload Type with Mixer |
RTP mixer modifies stream |
Payload type updated accordingly |
34 |
Payload Type with Clock Drift |
Sender clock drift |
Payload type unaffected |
35 |
Payload Type with Timestamp Mismatch |
Timestamp mismatch |
Payload type still identified |
36 |
Payload Type with High Bitrate |
High bitrate stream |
Payload type correctly parsed |
37 |
Payload Type with Low Bitrate |
Low bitrate stream |
Payload type correctly parsed |
38 |
Payload Type with Burst Traffic |
RTP packets in bursts |
Payload type correctly parsed |
39 |
Payload Type with Real-Time Monitoring |
RTP monitored live |
Payload type logged accurately |
40 |
Payload Type with Logging |
Payload type logged |
Correct payload type recorded |
41 |
Payload Type with SDP Mapping |
SDP maps dynamic payload |
Payload type resolved via SDP |
42 |
Payload Type with Codec Mismatch |
Payload type doesn’t match codec |
Error or warning generated |
43 |
Payload Type with Unknown Codec |
Unknown payload type |
Payload marked as unknown |
44 |
Payload Type with Multiple Codecs |
Multiple payload types in stream |
Each identified correctly |
45 |
Payload Type with RTP Profile |
RTP profile defines payloads |
Payload type resolved via profile |
46 |
Payload Type with Application Data |
Payload type used for app data |
Identified as application-specific |
47 |
Payload Type with Experimental Use |
Payload type in experimental range |
Identified as experimental |
48 |
Payload Type with Custom Codec |
Custom codec used |
Payload type resolved via signaling |
49 |
Payload Type with Interleaving |
Interleaved RTP packets |
Payload type correctly parsed |
50 |
Payload Type with Header Corruption |
Corrupted RTP header |
Payload type may be unreadable |
Source Identification - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
Valid SSRC |
RTP packet contains valid SSRC |
SSRC correctly identified |
2 |
Unique SSRC |
Each RTP stream has a unique SSRC |
Streams are distinguishable |
3 |
SSRC Collision |
Two sources use same SSRC |
Collision detected and resolved |
4 |
SSRC Change Mid-Stream |
SSRC changes during session |
New source identified |
5 |
SSRC with RTCP |
RTCP packet matches RTP SSRC |
Synchronization confirmed |
6 |
SSRC with Multiple Streams |
Multiple RTP streams with different SSRCs |
Each stream identified separately |
7 |
SSRC with Mixer |
RTP mixer assigns new SSRC |
Mixed stream identified correctly |
8 |
SSRC with Translator |
RTP translator preserves SSRC |
Original source traceable |
9 |
SSRC with Encryption |
Encrypted RTP packet |
SSRC decrypted and identified |
10 |
SSRC with SRTP |
Secure RTP used |
SSRC integrity verified |
11 |
SSRC with NAT |
RTP through NAT |
SSRC remains consistent |
12 |
SSRC with Firewall |
RTP through firewall |
SSRC correctly parsed |
13 |
SSRC with VPN |
RTP over VPN |
SSRC integrity maintained |
14 |
SSRC with IPv6 |
RTP over IPv6 |
SSRC correctly parsed |
15 |
SSRC with IPv4 |
RTP over IPv4 |
SSRC correctly parsed |
16 |
SSRC with Packet Loss |
Some packets lost |
SSRC still identifiable |
17 |
SSRC with Packet Reordering |
Packets arrive out of order |
SSRC correctly associated |
18 |
SSRC with Duplicate Packets |
Duplicate RTP packets |
SSRC remains consistent |
19 |
SSRC with Delay |
Delayed packets |
SSRC correctly parsed |
20 |
SSRC with Jitter |
Jitter in network |
SSRC unaffected |
21 |
SSRC with Timestamp Mismatch |
Timestamp mismatch |
SSRC still valid |
22 |
SSRC with Payload Change |
Payload type changes |
SSRC remains same |
23 |
SSRC with Marker Bit |
Marker bit set |
SSRC unaffected |
24 |
SSRC with Extension Header |
RTP extension present |
SSRC correctly parsed |
25 |
SSRC with RTCP BYE |
RTCP BYE received |
SSRC marked as inactive |
26 |
SSRC with RTCP SR |
RTCP Sender Report received |
SSRC matched with RTP |
27 |
SSRC with RTCP RR |
RTCP Receiver Report received |
SSRC matched with RTP |
28 |
SSRC with RTCP SDES |
RTCP SDES provides CNAME |
SSRC mapped to user |
29 |
SSRC with RTCP APP |
RTCP APP packet received |
SSRC correctly parsed |
30 |
SSRC with Multicast |
RTP over multicast |
SSRC identifies source |
31 |
SSRC with Unicast |
RTP over unicast |
SSRC identifies source |
32 |
SSRC with Clock Drift |
Sender clock drift |
SSRC unaffected |
33 |
SSRC with Session Restart |
RTP session restarted |
New SSRC assigned |
34 |
SSRC with Session Timeout |
No packets for timeout period |
SSRC marked as expired |
35 |
SSRC with High Bitrate |
High bitrate stream |
SSRC correctly parsed |
36 |
SSRC with Low Bitrate |
Low bitrate stream |
SSRC correctly parsed |
37 |
SSRC with Burst Traffic |
RTP packets in bursts |
SSRC correctly parsed |
38 |
SSRC with Silence Suppression |
Silence periods skipped |
SSRC continuity maintained |
39 |
SSRC with DTMF Events |
DTMF tones sent |
SSRC identifies source |
40 |
SSRC with Comfort Noise |
Comfort noise packets |
SSRC correctly parsed |
41 |
SSRC with FEC |
Forward Error Correction used |
SSRC identifies original stream |
42 |
SSRC with RED |
Redundant encoding used |
SSRC identifies original stream |
43 |
SSRC with Application Data |
Application-specific RTP |
SSRC identifies source |
44 |
SSRC with Logging |
SSRC logged for diagnostics |
SSRC recorded accurately |
45 |
SSRC with Real-Time Monitoring |
RTP monitored live |
SSRC used for stream tracking |
46 |
SSRC with SDP Mapping |
SSRC mapped via SDP |
SSRC resolved correctly |
47 |
SSRC with Unknown Source |
Unknown SSRC received |
New source identified |
48 |
SSRC with Multiple Codecs |
Multiple codecs in session |
SSRC used to separate streams |
49 |
SSRC with RTP Profile |
RTP profile defines behavior |
SSRC interpreted accordingly |
50 |
SSRC with Experimental Use |
Experimental RTP stream |
SSRC still uniquely identifies source |
Contribution Sourcr List - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
Validate single CSRC entry |
Send RTP packet with one CSRC |
CSRC list contains one SSRC |
2 |
Validate multiple CSRC entries |
Send RTP packet with multiple CSRCs |
All SSRCs appear in CSRC list |
3 |
Validate empty CSRC list |
Send RTP packet with no CSRC |
CSRC list is empty |
4 |
Validate max CSRC entries |
Send RTP packet with 15 CSRCs |
All 15 SSRCs are listed |
5 |
Validate CSRC with duplicate SSRCs |
Send RTP packet with repeated SSRCs |
Duplicates are ignored |
6 |
Validate CSRC with invalid SSRC |
Send RTP packet with malformed SSRC |
Packet is rejected or ignored |
7 |
Validate CSRC update on new source |
Add new SSRC mid-session |
New SSRC appears in CSRC list |
8 |
Validate CSRC removal on timeout |
Remove SSRC after inactivity |
Timed-out SSRC is removed |
9 |
Validate CSRC with mixed valid/invalid SSRCs |
Send RTP with both valid and invalid SSRCs |
Only valid SSRCs are listed |
10 |
Validate CSRC on SSRC collision |
Simulate SSRC collision |
Collision is handled gracefully |
11 |
Validate CSRC with RTP header extension |
Include CSRC and header extension |
CSRC is parsed correctly |
12 |
Validate CSRC with encrypted RTP |
Encrypt RTP stream |
CSRC remains accessible |
13 |
Validate CSRC with reordered packets |
Send RTP packets out of order |
CSRC list remains consistent |
14 |
Validate CSRC with packet loss |
Drop some RTP packets |
CSRC list remains accurate |
15 |
Validate CSRC with jitter |
Introduce jitter in RTP stream |
CSRC list remains stable |
16 |
Validate CSRC with retransmissions |
Retransmit RTP packets |
No duplicate SSRCs in CSRC list |
17 |
Validate CSRC with RTCP feedback |
Send RTCP reports |
CSRC list updates accordingly |
18 |
Validate CSRC on SSRC leave |
Remove SSRC from session |
SSRC is removed from CSRC list |
19 |
Validate CSRC on SSRC join |
Add SSRC to session |
SSRC is added to CSRC list |
20 |
Validate CSRC with malformed RTP |
Send corrupted RTP packet |
CSRC list is not updated |
21 |
Validate CSRC over TCP |
Send RTP over TCP |
CSRC list is maintained |
22 |
Validate CSRC over UDP |
Send RTP over UDP |
CSRC list is maintained |
23 |
Validate CSRC over TLS |
Send RTP over TLS |
CSRC list is maintained |
24 |
Validate CSRC over WebRTC |
Send RTP via WebRTC |
CSRC list is maintained |
25 |
Validate CSRC with RTP muxing |
Use RTP muxing |
CSRC identifies all sources |
26 |
Validate CSRC on SSRC switch |
Change SSRC mid-session |
CSRC list reflects new SSRC |
27 |
Validate CSRC on stream pause/resume |
Pause and resume RTP stream |
CSRC list remains consistent |
28 |
Validate CSRC on mute/unmute |
Mute and unmute RTP stream |
CSRC list remains consistent |
29 |
Validate CSRC on stream redirection |
Redirect RTP stream |
CSRC list updates accordingly |
30 |
Validate CSRC on stream duplication |
Duplicate RTP stream |
No duplicate SSRCs in CSRC list |
31 |
Validate CSRC from unknown source |
Send RTP from unknown SSRC |
Unknown SSRC is flagged |
32 |
Validate CSRC with IPv6 |
Send RTP from IPv6 SSRC |
CSRC list supports IPv6 |
33 |
Validate CSRC with IPv4 |
Send RTP from IPv4 SSRC |
CSRC list supports IPv4 |
34 |
Validate CSRC with delayed stream |
Delay RTP packets |
CSRC list updates after delay |
35 |
Validate CSRC with burst traffic |
Send RTP in bursts |
CSRC list handles burst correctly |
36 |
Validate CSRC behind NAT |
Send RTP from NATed SSRC |
CSRC list handles NAT traversal |
37 |
Validate CSRC over VPN |
Send RTP from VPN |
CSRC list handles VPN routing |
38 |
Validate CSRC on mobile network |
Send RTP from mobile |
CSRC list handles mobile jitter |
39 |
Validate CSRC on Wi-Fi |
Send RTP over Wi-Fi |
CSRC list handles Wi-Fi variability |
40 |
Validate CSRC on wired network |
Send RTP over LAN |
CSRC list handles wired stream |
41 |
Validate CSRC from satellite |
Send RTP from satellite |
CSRC list handles high latency |
42 |
Validate CSRC from multiple devices |
Send RTP from multiple devices |
CSRC aggregates all SSRCs |
43 |
Validate CSRC from same device, different apps |
Use two apps on same device |
CSRC distinguishes sources |
44 |
Validate CSRC from same app, different users |
Use same app with different users |
CSRC distinguishes users |
45 |
Validate CSRC with simulcast |
Send simulcast RTP streams |
CSRC handles simulcast correctly |
46 |
Validate CSRC with layered encoding |
Use SVC or similar |
CSRC handles layers correctly |
47 |
Validate CSRC from transcoder |
Send RTP from transcoder |
CSRC reflects transcoded SSRC |
48 |
Validate CSRC from mixer |
Send RTP from audio mixer |
CSRC includes all contributors |
49 |
Validate CSRC from relay |
Send RTP via relay |
CSRC reflects original SSRCs |
50 |
Validate CSRC from SFU |
Send RTP via SFU |
CSRC includes all forwarded SSRCs |
Header Extension - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
Validate presence of header extension |
Send RTP packet with header extension |
Header extension is detected |
2 |
Validate absence of header extension |
Send RTP packet without header extension |
No header extension is detected |
3 |
Validate multiple header extensions |
Send RTP packet with multiple header extensions |
All header extensions are parsed |
4 |
Validate header extension with valid ID |
Use a valid extension ID in RTP header |
Extension is accepted |
5 |
Validate header extension with invalid ID |
Use an invalid extension ID |
Extension is ignored or rejected |
6 |
Validate header extension with maximum length |
Send extension with maximum allowed length |
Extension is parsed correctly |
7 |
Validate header extension with zero length |
Send extension with zero length |
Extension is ignored |
8 |
Validate header extension with padding |
Send extension with padding bytes |
Padding is handled correctly |
9 |
Validate header extension with one-byte format |
Use one-byte header extension format |
Extension is parsed correctly |
10 |
Validate header extension with two-byte format |
Use two-byte header extension format |
Extension is parsed correctly |
11 |
Validate header extension parsing order |
Send multiple extensions in specific order |
Order is preserved |
12 |
Validate header extension with unknown ID |
Send extension with unknown ID |
Extension is ignored |
13 |
Validate header extension with encrypted RTP |
Send encrypted RTP with extension |
Extension is still accessible |
14 |
Validate header extension with reordered packets |
Send RTP packets out of order |
Extensions are parsed correctly |
15 |
Validate header extension with packet loss |
Drop some RTP packets |
Extensions in received packets are parsed |
16 |
Validate header extension with jitter |
Introduce jitter in RTP stream |
Extensions are parsed correctly |
17 |
Validate header extension with retransmissions |
Retransmit RTP packets |
No duplicate extensions are parsed |
18 |
Validate header extension with RTCP feedback |
Send RTCP reports |
Extensions are unaffected |
19 |
Validate header extension with malformed data |
Send extension with malformed content |
Extension is ignored |
20 |
Validate header extension with RTP over TCP |
Send RTP over TCP |
Extensions are parsed correctly |
21 |
Validate header extension with RTP over UDP |
Send RTP over UDP |
Extensions are parsed correctly |
22 |
Validate header extension with RTP over TLS |
Send RTP over TLS |
Extensions are parsed correctly |
23 |
Validate header extension with RTP over WebRTC |
Send RTP via WebRTC |
Extensions are parsed correctly |
24 |
Validate header extension with muxed RTP |
Use RTP muxing |
Extensions are parsed correctly |
25 |
Validate header extension with SSRC change |
Change SSRC mid-session |
Extensions are still parsed |
26 |
Validate header extension with stream pause/resume |
Pause and resume RTP stream |
Extensions are parsed correctly |
27 |
Validate header extension with mute/unmute |
Mute and unmute RTP stream |
Extensions are parsed correctly |
28 |
Validate header extension with stream redirection |
Redirect RTP stream |
Extensions are parsed correctly |
29 |
Validate header extension with stream duplication |
Duplicate RTP stream |
No duplicate extensions are parsed |
30 |
Validate header extension from unknown source |
Send RTP from unknown SSRC |
Extensions are parsed correctly |
31 |
Validate header extension with IPv6 |
Send RTP from IPv6 source |
Extensions are parsed correctly |
32 |
Validate header extension with IPv4 |
Send RTP from IPv4 source |
Extensions are parsed correctly |
33 |
Validate header extension with delayed stream |
Delay RTP packets |
Extensions are parsed correctly |
34 |
Validate header extension with burst traffic |
Send RTP in bursts |
Extensions are parsed correctly |
35 |
Validate header extension behind NAT |
Send RTP from NATed source |
Extensions are parsed correctly |
36 |
Validate header extension over VPN |
Send RTP from VPN |
Extensions are parsed correctly |
37 |
Validate header extension on mobile network |
Send RTP from mobile |
Extensions are parsed correctly |
38 |
Validate header extension on Wi-Fi |
Send RTP over Wi-Fi |
Extensions are parsed correctly |
39 |
Validate header extension on wired network |
Send RTP over LAN |
Extensions are parsed correctly |
40 |
Validate header extension from satellite |
Send RTP from satellite |
Extensions are parsed correctly |
41 |
Validate header extension from multiple devices |
Send RTP from multiple devices |
Extensions are parsed correctly |
42 |
Validate header extension from same device, different apps |
Use two apps on same device |
Extensions are parsed correctly |
43 |
Validate header extension from same app, different users |
Use same app with different users |
Extensions are parsed correctly |
44 |
Validate header extension with simulcast |
Send simulcast RTP streams |
Extensions are parsed correctly |
45 |
Validate header extension with layered encoding |
Use SVC or similar |
Extensions are parsed correctly |
46 |
Validate header extension from transcoder |
Send RTP from transcoder |
Extensions are parsed correctly |
47 |
Validate header extension from mixer |
Send RTP from audio mixer |
Extensions are parsed correctly |
48 |
Validate header extension from relay |
Send RTP via relay |
Extensions are parsed correctly |
49 |
Validate header extension from SFU |
Send RTP via SFU |
Extensions are parsed correctly |
50 |
Validate header extension with dynamic update |
Update extension mid-session |
New extension is parsed correctly |
Support for Multicast and Unicast - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
Test basic multicast stream |
Send RTP stream to multicast group |
All group members receive stream |
2 |
Test basic unicast stream |
Send RTP stream to single client |
Only target client receives stream |
3 |
Test multicast join |
Client joins multicast group |
Client starts receiving RTP packets |
4 |
Test multicast leave |
Client leaves multicast group |
Client stops receiving RTP packets |
5 |
Test unicast to multiple clients |
Send unicast to multiple clients |
Each client receives separate stream |
6 |
Test multicast with IGMPv2 |
Join multicast using IGMPv2 |
Stream is received successfully |
7 |
Test multicast with IGMPv3 |
Join multicast using IGMPv3 |
Stream is received successfully |
8 |
Test multicast with TTL = 1 |
Send multicast with TTL=1 |
Only local subnet receives stream |
9 |
Test multicast with TTL = 0 |
Send multicast with TTL=0 |
Stream is not forwarded |
10 |
Test multicast with TTL = 255 |
Send multicast with TTL=255 |
Stream reaches all allowed hops |
11 |
Test unicast with NAT |
Send unicast through NAT |
Stream is received with NAT traversal |
12 |
Test multicast with NAT |
Send multicast through NAT |
Stream is blocked or dropped |
13 |
Test multicast with firewall |
Block multicast port |
Stream is not received |
14 |
Test unicast with firewall |
Allow unicast port |
Stream is received |
15 |
Test multicast with packet loss |
Simulate packet loss |
Stream quality degrades |
16 |
Test unicast with packet loss |
Simulate packet loss |
Stream quality degrades |
17 |
Test multicast with jitter |
Introduce jitter |
Stream is received with delay |
18 |
Test unicast with jitter |
Introduce jitter |
Stream is received with delay |
19 |
Test multicast with encryption |
Encrypt RTP stream |
Stream is received and decrypted |
20 |
Test unicast with encryption |
Encrypt RTP stream |
Stream is received and decrypted |
21 |
Test multicast with RTCP |
Send RTCP reports |
Reports are aggregated |
22 |
Test unicast with RTCP |
Send RTCP reports |
Reports are per client |
23 |
Test multicast with IPv6 |
Send multicast over IPv6 |
Stream is received |
24 |
Test unicast with IPv6 |
Send unicast over IPv6 |
Stream is received |
25 |
Test multicast with IPv4 |
Send multicast over IPv4 |
Stream is received |
26 |
Test unicast with IPv4 |
Send unicast over IPv4 |
Stream is received |
27 |
Test multicast with multiple sources |
Send from multiple sources |
All sources are received |
28 |
Test unicast with multiple sources |
Send from multiple sources |
All streams are received |
29 |
Test multicast with stream pause |
Pause stream |
Clients stop receiving |
30 |
Test unicast with stream pause |
Pause stream |
Client stops receiving |
31 |
Test multicast with stream resume |
Resume stream |
Clients resume receiving |
32 |
Test unicast with stream resume |
Resume stream |
Client resumes receiving |
33 |
Test multicast with stream redirection |
Redirect stream |
Clients receive from new source |
34 |
Test unicast with stream redirection |
Redirect stream |
Client receives from new source |
35 |
Test multicast with burst traffic |
Send burst packets |
Clients handle burst |
36 |
Test unicast with burst traffic |
Send burst packets |
Client handles burst |
37 |
Test multicast with mobile client |
Join from mobile |
Stream is received |
38 |
Test unicast with mobile client |
Send to mobile |
Stream is received |
39 |
Test multicast with Wi-Fi |
Join from Wi-Fi |
Stream is received |
40 |
Test unicast with Wi-Fi |
Send to Wi-Fi |
Stream is received |
41 |
Test multicast with wired LAN |
Join from LAN |
Stream is received |
42 |
Test unicast with wired LAN |
Send to LAN |
Stream is received |
43 |
Test multicast with VPN |
Join from VPN |
Stream may be blocked |
44 |
Test unicast with VPN |
Send to VPN |
Stream is received |
45 |
Test multicast with SFU |
Use SFU to forward multicast |
Stream is forwarded |
46 |
Test unicast with SFU |
Use SFU to forward unicast |
Stream is forwarded |
47 |
Test multicast with relay |
Use relay for multicast |
Stream is forwarded |
48 |
Test unicast with relay |
Use relay for unicast |
Stream is forwarded |
49 |
Test multicast with congestion |
Simulate network congestion |
Stream quality degrades |
50 |
Test unicast with congestion |
Simulate network congestion |
Stream quality degrades |
Works with RTCP - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
Validate RTCP Sender Report |
Send RTP with RTCP SR |
Receiver receives valid SR |
2 |
Validate RTCP Receiver Report |
Send RTP with RTCP RR |
Sender receives valid RR |
3 |
Validate RTCP SDES packet |
Send RTCP SDES info |
SDES info is parsed correctly |
4 |
Validate RTCP BYE packet |
Send RTCP BYE |
Session ends gracefully |
5 |
Validate RTCP APP packet |
Send RTCP APP packet |
Application-specific data is handled |
6 |
Validate RTCP interval timing |
Check RTCP interval |
RTCP packets sent at correct intervals |
7 |
Validate RTCP with encryption |
Encrypt RTCP packets |
RTCP is decrypted and parsed |
8 |
Validate RTCP with packet loss |
Simulate RTP loss |
RTCP reports reflect loss |
9 |
Validate RTCP with jitter |
Introduce jitter |
RTCP reports reflect jitter |
10 |
Validate RTCP with delay |
Delay RTP packets |
RTCP reports reflect delay |
11 |
Validate RTCP with multiple SSRCs |
Use multiple sources |
RTCP handles all SSRCs |
12 |
Validate RTCP with SSRC collision |
Simulate SSRC conflict |
RTCP resolves collision |
13 |
Validate RTCP with NAT traversal |
Send through NAT |
RTCP reaches destination |
14 |
Validate RTCP with firewall |
Block RTCP port |
RTCP is dropped or rerouted |
15 |
Validate RTCP with IPv4 |
Use IPv4 transport |
RTCP is transmitted correctly |
16 |
Validate RTCP with IPv6 |
Use IPv6 transport |
RTCP is transmitted correctly |
17 |
Validate RTCP with RTP muxing |
Use RTP/RTCP mux |
RTCP is demuxed correctly |
18 |
Validate RTCP with SRTP |
Use secure RTP |
RTCP is secured and parsed |
19 |
Validate RTCP with RTCP-XR |
Send extended reports |
RTCP-XR is parsed correctly |
20 |
Validate RTCP with mobile network |
Use mobile data |
RTCP is transmitted reliably |
21 |
Validate RTCP with Wi-Fi |
Use Wi-Fi |
RTCP is transmitted reliably |
22 |
Validate RTCP with wired LAN |
Use Ethernet |
RTCP is transmitted reliably |
23 |
Validate RTCP with VPN |
Use VPN tunnel |
RTCP is transmitted securely |
24 |
Validate RTCP with SFU |
Use SFU in path |
RTCP is forwarded correctly |
25 |
Validate RTCP with relay |
Use relay server |
RTCP is relayed correctly |
26 |
Validate RTCP with simulcast |
Use simulcast streams |
RTCP handles all layers |
27 |
Validate RTCP with SVC |
Use scalable video |
RTCP handles all layers |
28 |
Validate RTCP with burst traffic |
Send RTP bursts |
RTCP reflects burst stats |
29 |
Validate RTCP with muted stream |
Mute RTP stream |
RTCP still reports session |
30 |
Validate RTCP with paused stream |
Pause RTP stream |
RTCP continues reporting |
31 |
Validate RTCP with resumed stream |
Resume RTP stream |
RTCP reflects resumed stats |
32 |
Validate RTCP with stream redirection |
Redirect RTP |
RTCP follows new path |
33 |
Validate RTCP with stream duplication |
Duplicate RTP |
RTCP handles duplicates |
34 |
Validate RTCP with unknown SSRC |
Send unknown SSRC |
RTCP ignores or flags it |
35 |
Validate RTCP with malformed packet |
Corrupt RTCP packet |
Packet is dropped |
36 |
Validate RTCP with large payload |
Send large RTCP packet |
Packet is handled correctly |
37 |
Validate RTCP with minimal payload |
Send minimal RTCP |
Packet is accepted |
38 |
Validate RTCP with RTSP |
Use RTSP signaling |
RTCP is integrated |
39 |
Validate RTCP with SIP |
Use SIP signaling |
RTCP is integrated |
40 |
Validate RTCP with congestion |
Simulate congestion |
RTCP reflects network stats |
41 |
Validate RTCP with QoS monitoring |
Enable QoS |
RTCP provides metrics |
42 |
Validate RTCP with timestamp drift |
Introduce drift |
RTCP reports sync issues |
43 |
Validate RTCP with clock skew |
Simulate skew |
RTCP compensates |
44 |
Validate RTCP with RTP retransmission |
Enable RTX |
RTCP reflects retransmits |
45 |
Validate RTCP with feedback messages |
Use RTCP feedback |
Feedback is parsed |
46 |
Validate RTCP with NACK |
Send NACK |
Packet loss is reported |
47 |
Validate RTCP with PLI |
Send PLI |
Keyframe is requested |
48 |
Validate RTCP with FIR |
Send FIR |
Full intra-frame is requested |
49 |
Validate RTCP with REMB |
Send REMB |
Bandwidth is adjusted |
50 |
Validate RTCP with TMMBR |
Send TMMBR |
Bitrate is adjusted |
Codec Agnostic - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
Validate RTP with H.264 codec |
Send RTP stream using H.264 |
RTP stream is transmitted successfully |
2 |
Validate RTP with VP8 codec |
Send RTP stream using VP8 |
RTP stream is transmitted successfully |
3 |
Validate RTP with VP9 codec |
Send RTP stream using VP9 |
RTP stream is transmitted successfully |
4 |
Validate RTP with AV1 codec |
Send RTP stream using AV1 |
RTP stream is transmitted successfully |
5 |
Validate RTP with Opus codec |
Send RTP stream using Opus |
RTP stream is transmitted successfully |
6 |
Validate RTP with G.711 codec |
Send RTP stream using G.711 |
RTP stream is transmitted successfully |
7 |
Validate RTP with G.729 codec |
Send RTP stream using G.729 |
RTP stream is transmitted successfully |
8 |
Validate RTP with AAC codec |
Send RTP stream using AAC |
RTP stream is transmitted successfully |
9 |
Validate RTP with MP3 codec |
Send RTP stream using MP3 |
RTP stream is transmitted successfully |
10 |
Validate RTP with PCM codec |
Send RTP stream using PCM |
RTP stream is transmitted successfully |
11 |
Validate RTP with unknown codec |
Send RTP stream with unknown codec |
RTP stream is transmitted without codec dependency |
12 |
Validate RTP with codec switch mid-session |
Switch codec during session |
RTP continues without interruption |
13 |
Validate RTP with multiple codecs |
Use multiple codecs in session |
RTP handles all codecs |
14 |
Validate RTP with codec negotiation |
Perform codec negotiation |
RTP adapts to selected codec |
15 |
Validate RTP with codec mismatch |
Use different codecs on sender/receiver |
RTP stream is transmitted, decoding may fail |
16 |
Validate RTP with codec fallback |
Fallback to secondary codec |
RTP stream continues |
17 |
Validate RTP with codec upgrade |
Upgrade codec mid-session |
RTP stream continues |
18 |
Validate RTP with codec downgrade |
Downgrade codec mid-session |
RTP stream continues |
19 |
Validate RTP with codec-specific payload |
Send codec-specific payload |
RTP handles payload correctly |
20 |
Validate RTP with dynamic payload type |
Use dynamic payload types |
RTP stream is transmitted successfully |
21 |
Validate RTP with static payload type |
Use static payload types |
RTP stream is transmitted successfully |
22 |
Validate RTP with codec over UDP |
Send RTP over UDP |
RTP stream is codec agnostic |
23 |
Validate RTP with codec over TCP |
Send RTP over TCP |
RTP stream is codec agnostic |
24 |
Validate RTP with codec over TLS |
Send RTP over TLS |
RTP stream is codec agnostic |
25 |
Validate RTP with codec over WebRTC |
Send RTP via WebRTC |
RTP stream is codec agnostic |
26 |
Validate RTP with codec over SRTP |
Send RTP via SRTP |
RTP stream is codec agnostic |
27 |
Validate RTP with codec over VPN |
Send RTP via VPN |
RTP stream is codec agnostic |
28 |
Validate RTP with codec over NAT |
Send RTP through NAT |
RTP stream is codec agnostic |
29 |
Validate RTP with codec over firewall |
Send RTP through firewall |
RTP stream is codec agnostic |
30 |
Validate RTP with codec over mobile network |
Send RTP via mobile |
RTP stream is codec agnostic |
31 |
Validate RTP with codec over Wi-Fi |
Send RTP via Wi-Fi |
RTP stream is codec agnostic |
32 |
Validate RTP with codec over LAN |
Send RTP via LAN |
RTP stream is codec agnostic |
33 |
Validate RTP with codec over satellite |
Send RTP via satellite |
RTP stream is codec agnostic |
34 |
Validate RTP with codec under jitter |
Introduce jitter |
RTP stream is codec agnostic |
35 |
Validate RTP with codec under packet loss |
Simulate packet loss |
RTP stream is codec agnostic |
36 |
Validate RTP with codec under delay |
Introduce delay |
RTP stream is codec agnostic |
37 |
Validate RTP with codec under congestion |
Simulate congestion |
RTP stream is codec agnostic |
38 |
Validate RTP with codec in multicast |
Send RTP via multicast |
RTP stream is codec agnostic |
39 |
Validate RTP with codec in unicast |
Send RTP via unicast |
RTP stream is codec agnostic |
40 |
Validate RTP with codec in SFU |
Use SFU for forwarding |
RTP stream is codec agnostic |
41 |
Validate RTP with codec in relay |
Use relay server |
RTP stream is codec agnostic |
42 |
Validate RTP with codec in mixer |
Use media mixer |
RTP stream is codec agnostic |
43 |
Validate RTP with codec in transcoder |
Use transcoder |
RTP stream is codec agnostic |
44 |
Validate RTP with codec in recording |
Record RTP stream |
RTP stream is codec agnostic |
45 |
Validate RTP with codec in playback |
Play RTP stream |
RTP stream is codec agnostic |
46 |
Validate RTP with codec in conferencing |
Use in video conference |
RTP stream is codec agnostic |
47 |
Validate RTP with codec in broadcasting |
Use in live broadcast |
RTP stream is codec agnostic |
48 |
Validate RTP with codec in surveillance |
Use in CCTV system |
RTP stream is codec agnostic |
49 |
Validate RTP with codec in gaming |
Use in game streaming |
RTP stream is codec agnostic |
50 |
Validate RTP with codec in education |
Use in e-learning |
RTP stream is codec agnostic |
Transport Flexibility - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
RTP over UDP |
Send RTP stream over UDP |
RTP packets are transmitted successfully |
2 |
RTP over TCP |
Send RTP stream over TCP |
RTP packets are transmitted successfully |
3 |
RTP over TLS |
Send RTP stream over TLS |
RTP packets are encrypted and transmitted |
4 |
RTP over WebRTC |
Send RTP stream via WebRTC |
RTP packets are transmitted with WebRTC signaling |
5 |
RTP over QUIC |
Send RTP stream over QUIC |
RTP packets are transmitted with low latency |
6 |
RTP over SCTP |
Send RTP stream over SCTP |
RTP packets are transmitted reliably |
7 |
RTP over HTTP/2 |
Send RTP stream over HTTP/2 |
RTP packets are tunneled successfully |
8 |
RTP over VPN |
Send RTP stream through VPN |
RTP packets are transmitted securely |
9 |
RTP over NAT |
Send RTP stream through NAT |
RTP packets traverse NAT successfully |
10 |
RTP over firewall |
Send RTP stream through firewall |
RTP packets are allowed through firewall rules |
11 |
RTP over proxy |
Send RTP stream via proxy server |
RTP packets are routed correctly |
12 |
RTP over mobile network |
Send RTP stream via 4G/5G |
RTP packets are transmitted with adaptive bitrate |
13 |
RTP over satellite |
Send RTP stream via satellite link |
RTP packets are transmitted with high latency |
14 |
RTP over Wi-Fi |
Send RTP stream over Wi-Fi |
RTP packets are transmitted with variable jitter |
15 |
RTP over Ethernet |
Send RTP stream over wired LAN |
RTP packets are transmitted with low jitter |
16 |
RTP over IPv4 |
Send RTP stream using IPv4 |
RTP packets are transmitted successfully |
17 |
RTP over IPv6 |
Send RTP stream using IPv6 |
RTP packets are transmitted successfully |
18 |
RTP over dual stack |
Send RTP stream using IPv4/IPv6 dual stack |
RTP packets are transmitted over both protocols |
19 |
RTP with transport switching |
Switch transport mid-session |
RTP stream continues without interruption |
20 |
RTP with fallback transport |
Fallback to alternate transport on failure |
RTP stream resumes on fallback transport |
21 |
RTP with multiplexing |
Use RTP multiplexing over single port |
Multiple RTP streams are transmitted successfully |
22 |
RTP with demultiplexing |
Separate RTP streams from multiplexed transport |
Streams are correctly identified |
23 |
RTP with congestion control |
Simulate network congestion |
RTP adapts bitrate and transport |
24 |
RTP with packet loss recovery |
Simulate packet loss |
RTP uses retransmission or FEC |
25 |
RTP with jitter buffering |
Introduce jitter |
RTP stream is smoothed by buffer |
26 |
RTP with encryption |
Encrypt RTP stream |
RTP packets are secure and readable |
27 |
RTP with authentication |
Authenticate RTP source |
RTP packets are verified |
28 |
RTP with transport header extension |
Add transport-specific header |
RTP packets are parsed correctly |
29 |
RTP with transport QoS tagging |
Tag RTP packets with QoS |
Network prioritizes RTP traffic |
30 |
RTP with DSCP marking |
Mark RTP packets with DSCP |
RTP packets are prioritized in network |
31 |
RTP with SRTP over UDP |
Send SRTP over UDP |
Encrypted RTP packets are transmitted |
32 |
RTP with SRTP over TCP |
Send SRTP over TCP |
Encrypted RTP packets are transmitted |
33 |
RTP with SRTP over TLS |
Send SRTP over TLS |
Encrypted RTP packets are secure |
34 |
RTP with DTLS transport |
Use DTLS for RTP transport |
RTP packets are encrypted and transmitted |
35 |
RTP with ICE negotiation |
Use ICE for transport selection |
RTP transport is negotiated successfully |
36 |
RTP with STUN traversal |
Use STUN for NAT traversal |
RTP packets traverse NAT |
37 |
RTP with TURN relay |
Use TURN server for transport |
RTP packets are relayed successfully |
38 |
RTP with transport redundancy |
Use redundant transport paths |
RTP stream is resilient to failure |
39 |
RTP with transport monitoring |
Monitor transport metrics |
RTP transport stats are collected |
40 |
RTP with transport adaptation |
Adapt transport based on network |
RTP switches to optimal transport |
41 |
RTP with transport error handling |
Simulate transport error |
RTP handles error gracefully |
42 |
RTP with transport timeout |
Simulate transport timeout |
RTP retries or switches transport |
43 |
RTP with transport delay |
Introduce transport delay |
RTP stream compensates for delay |
44 |
RTP with transport burst |
Send RTP in bursts |
RTP stream handles burst traffic |
45 |
RTP with transport fragmentation |
Fragment RTP packets |
RTP packets are reassembled correctly |
46 |
RTP with transport aggregation |
Aggregate RTP packets |
RTP packets are transmitted efficiently |
47 |
RTP with transport prioritization |
Prioritize RTP over other traffic |
RTP packets are transmitted with priority |
48 |
RTP with transport scheduling |
Schedule RTP transmission |
RTP packets follow schedule |
49 |
RTP with transport logging |
Log transport activity |
RTP transport logs are generated |
50 |
RTP with transport simulation |
Simulate various transport types |
RTP adapts to simulated conditions |
Extensibility - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
Validate custom RTP header extension |
Add custom header field |
RTP packet includes custom field |
2 |
Validate dynamic payload type support |
Use dynamic payload types |
RTP handles them correctly |
3 |
Validate new codec integration |
Add new codec to RTP stream |
RTP transmits without issues |
4 |
Validate RTP with new transport protocol |
Use RTP over QUIC |
RTP adapts to new transport |
5 |
Validate RTP with new encryption scheme |
Use custom encryption |
RTP packets are encrypted/decrypted |
6 |
Validate RTP with new RTCP feedback |
Add new RTCP feedback type |
RTP handles feedback correctly |
7 |
Validate RTP with new media type |
Add support for 3D video |
RTP transmits new media type |
8 |
Validate RTP with new clock rate |
Use non-standard clock rate |
RTP timestamps are accurate |
9 |
Validate RTP with new synchronization method |
Use custom sync method |
RTP syncs streams correctly |
10 |
Validate RTP with new jitter buffer algorithm |
Implement custom jitter buffer |
RTP playback is smooth |
11 |
Validate RTP with new congestion control |
Add new congestion control logic |
RTP adapts to network conditions |
12 |
Validate RTP with new FEC mechanism |
Add custom FEC |
RTP recovers lost packets |
13 |
Validate RTP with new retransmission logic |
Implement custom RTX |
RTP handles retransmissions |
14 |
Validate RTP with new multiplexing scheme |
Use new muxing logic |
RTP demuxes correctly |
15 |
Validate RTP with new header compression |
Add custom compression |
RTP headers are compressed |
16 |
Validate RTP with new signaling protocol |
Use custom signaling |
RTP session is established |
17 |
Validate RTP with new NAT traversal method |
Use custom NAT traversal |
RTP reaches peer |
18 |
Validate RTP with new QoS tagging |
Add new DSCP values |
RTP packets are prioritized |
19 |
Validate RTP with new timestamp format |
Use extended timestamp |
RTP syncs correctly |
20 |
Validate RTP with new SSRC allocation |
Use custom SSRC logic |
RTP identifies sources |
21 |
Validate RTP with new media switching logic |
Implement codec switching |
RTP switches seamlessly |
22 |
Validate RTP with new error correction |
Add new error correction |
RTP handles errors |
23 |
Validate RTP with new stream prioritization |
Prioritize streams |
RTP sends high-priority first |
24 |
Validate RTP with new payload format |
Define new payload format |
RTP parses correctly |
25 |
Validate RTP with new transport profile |
Use custom RTP profile |
RTP conforms to profile |
26 |
Validate RTP with new encryption key exchange |
Use custom key exchange |
RTP secures stream |
27 |
Validate RTP with new authentication method |
Add custom auth |
RTP verifies packets |
28 |
Validate RTP with new session control |
Use custom session control |
RTP session is managed |
29 |
Validate RTP with new media adaptation |
Adapt to resolution change |
RTP adjusts stream |
30 |
Validate RTP with new stream labeling |
Add stream labels |
RTP identifies streams |
31 |
Validate RTP with new metadata format |
Add custom metadata |
RTP includes metadata |
32 |
Validate RTP with new packet pacing |
Implement pacing logic |
RTP avoids congestion |
33 |
Validate RTP with new timestamp wraparound handling |
Handle wraparound |
RTP syncs correctly |
34 |
Validate RTP with new stream repair logic |
Add repair packets |
RTP recovers stream |
35 |
Validate RTP with new stream monitoring |
Add monitoring hooks |
RTP stats are collected |
36 |
Validate RTP with new stream analytics |
Add analytics module |
RTP usage is analyzed |
37 |
Validate RTP with new stream recording |
Add recording logic |
RTP is stored |
38 |
Validate RTP with new stream playback |
Add playback support |
RTP is replayed |
39 |
Validate RTP with new stream mixing |
Mix multiple streams |
RTP outputs mixed stream |
40 |
Validate RTP with new stream splitting |
Split stream to clients |
RTP delivers to all |
41 |
Validate RTP with new stream forwarding |
Forward stream via proxy |
RTP reaches destination |
42 |
Validate RTP with new stream duplication |
Duplicate stream |
RTP sends to multiple targets |
43 |
Validate RTP with new stream filtering |
Filter by codec |
RTP sends only selected |
44 |
Validate RTP with new stream tagging |
Tag packets |
RTP includes tags |
45 |
Validate RTP with new stream scheduling |
Schedule streams |
RTP sends on schedule |
46 |
Validate RTP with new stream prioritization |
Prioritize by user |
RTP sends VIP stream first |
47 |
Validate RTP with new stream encryption per user |
Encrypt per user |
RTP secures each stream |
48 |
Validate RTP with new stream routing |
Route by region |
RTP reaches correct region |
49 |
Validate RTP with new stream adaptation to bandwidth |
Adjust bitrate |
RTP adapts stream |
50 |
Validate RTP with new stream format negotiation |
Negotiate format |
RTP uses agreed format |
Interoperability - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
RTP between different vendors |
Send RTP between two vendor systems |
RTP stream is received correctly |
2 |
RTP between different codecs |
Use different codecs on sender and receiver |
RTP stream is decoded correctly |
3 |
RTP between IPv4 and IPv6 |
Send RTP from IPv4 to IPv6 system |
RTP stream is transmitted successfully |
4 |
RTP between mobile and desktop |
Send RTP from mobile to desktop |
RTP stream is interoperable |
5 |
RTP between Linux and Windows |
Send RTP between Linux and Windows systems |
RTP stream is received correctly |
6 |
RTP between browser and native app |
Send RTP from browser to native app |
RTP stream is compatible |
7 |
RTP between WebRTC and SIP |
Send RTP between WebRTC and SIP endpoints |
RTP stream is interoperable |
8 |
RTP between encrypted and unencrypted streams |
Mix SRTP and RTP |
Encrypted stream is handled correctly |
9 |
RTP with different payload types |
Use different payload types |
RTP stream is interpreted correctly |
10 |
RTP with different clock rates |
Use different clock rates |
RTP stream is synchronized |
11 |
RTP with different sampling rates |
Use different audio sampling rates |
RTP stream is decoded correctly |
12 |
RTP with different packet sizes |
Use varying packet sizes |
RTP stream is processed correctly |
13 |
RTP with different transport protocols |
Use UDP and TCP |
RTP stream is interoperable |
14 |
RTP with different jitter buffers |
Use different jitter buffer sizes |
RTP stream is smooth |
15 |
RTP with different RTCP implementations |
Use different RTCP formats |
RTCP feedback is interpreted correctly |
16 |
RTP with different header extensions |
Use custom header extensions |
Extensions are parsed correctly |
17 |
RTP with different SSRC formats |
Use different SSRC values |
RTP stream is identified correctly |
18 |
RTP with different CSRC lists |
Use different CSRC configurations |
RTP stream is processed correctly |
19 |
RTP with different media types |
Use audio and video RTP streams |
Both streams are handled correctly |
20 |
RTP with different synchronization sources |
Use different sync sources |
RTP stream is synchronized |
21 |
RTP with different session descriptions |
Use different SDP formats |
RTP session is established correctly |
22 |
RTP with different NAT traversal methods |
Use STUN, TURN, ICE |
RTP stream is transmitted successfully |
23 |
RTP with different encryption methods |
Use DTLS, SRTP |
RTP stream is secure and interoperable |
24 |
RTP with different authentication methods |
Use different auth schemes |
RTP stream is verified |
25 |
RTP with different media servers |
Use different RTP media servers |
RTP stream is routed correctly |
26 |
RTP with different conferencing platforms |
Use Zoom, Teams, Webex |
RTP stream is interoperable |
27 |
RTP with different network conditions |
Simulate varying network conditions |
RTP stream adapts correctly |
28 |
RTP with different bandwidths |
Use low and high bandwidth |
RTP stream adjusts bitrate |
29 |
RTP with different codecs in same session |
Use multiple codecs |
RTP stream is decoded correctly |
30 |
RTP with transcoding |
Transcode RTP stream mid-session |
RTP stream is received correctly |
31 |
RTP with simulcast |
Use simulcast RTP streams |
All streams are received correctly |
32 |
RTP with layered encoding |
Use SVC or similar |
RTP layers are decoded correctly |
33 |
RTP with different RTP profiles |
Use AVP, SAVP |
RTP stream is interpreted correctly |
34 |
RTP with different media formats |
Use H.264, VP8, Opus |
RTP stream is decoded correctly |
35 |
RTP with different timing models |
Use different timestamp models |
RTP stream is synchronized |
36 |
RTP with different buffer strategies |
Use adaptive buffering |
RTP stream is smooth |
37 |
RTP with different session control protocols |
Use SIP, H.323 |
RTP stream is interoperable |
38 |
RTP with different firewall configurations |
Use strict and open firewalls |
RTP stream is transmitted successfully |
39 |
RTP with different QoS policies |
Use different QoS settings |
RTP stream is prioritized correctly |
40 |
RTP with different DSCP markings |
Use different DSCP values |
RTP stream is routed correctly |
41 |
RTP with different media gateways |
Use different RTP gateways |
RTP stream is forwarded correctly |
42 |
RTP with different RTP libraries |
Use different RTP implementations |
RTP stream is interoperable |
43 |
RTP with different RTP versions |
Use RTP 1.0 and RTP 2.0 |
RTP stream is compatible |
44 |
RTP with different RTP extensions |
Use custom RTP extensions |
RTP stream is parsed correctly |
45 |
RTP with different session negotiation methods |
Use manual and automatic negotiation |
RTP session is established |
46 |
RTP with different stream directions |
Use sendonly, recvonly, sendrecv |
RTP stream is handled correctly |
47 |
RTP with different media capabilities |
Use different codec capabilities |
RTP stream is negotiated correctly |
48 |
RTP with different stream priorities |
Use priority tags |
RTP stream is prioritized |
49 |
RTP with different stream identifiers |
Use different stream IDs |
RTP stream is identified correctly |
50 |
RTP with different error handling mechanisms |
Use retry, fallback |
RTP stream is resilient |
Security via SRTP - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
SRTP encryption enabled |
Enable SRTP on RTP stream |
RTP packets are encrypted |
2 |
SRTP decryption at receiver |
Decrypt SRTP packets at receiver |
RTP stream is decoded correctly |
3 |
SRTP with AES-128 encryption |
Use AES-128 for SRTP |
Packets are encrypted with AES-128 |
4 |
SRTP with AES-256 encryption |
Use AES-256 for SRTP |
Packets are encrypted with AES-256 |
5 |
SRTP with HMAC-SHA1 authentication |
Use HMAC-SHA1 for integrity |
Packets are authenticated |
6 |
SRTP with NULL cipher |
Use NULL cipher for testing |
Packets are transmitted unencrypted |
7 |
SRTP with replay protection |
Enable replay protection |
Duplicate packets are discarded |
8 |
SRTP with key rollover |
Rotate encryption keys mid-session |
Stream continues securely |
9 |
SRTP with DTLS key exchange |
Use DTLS for key negotiation |
Keys are exchanged securely |
10 |
SRTP with static key |
Use pre-shared static key |
Stream is encrypted and decrypted |
11 |
SRTP with dynamic key |
Use dynamic key exchange |
Stream is encrypted and decrypted |
12 |
SRTP with key mismatch |
Use incorrect key at receiver |
Decryption fails |
13 |
SRTP with packet loss |
Simulate packet loss |
Stream remains secure |
14 |
SRTP with jitter |
Introduce jitter |
Stream remains secure |
15 |
SRTP with out-of-order packets |
Send packets out of order |
Stream is decrypted correctly |
16 |
SRTP with malformed packets |
Send corrupted SRTP packets |
Packets are rejected |
17 |
SRTP with header extension |
Include RTP header extension |
Extension is encrypted correctly |
18 |
SRTP with RTCP encryption |
Encrypt RTCP packets |
RTCP packets are secure |
19 |
SRTP with RTCP feedback |
Send encrypted RTCP feedback |
Feedback is received securely |
20 |
SRTP with SSRC collision |
Simulate SSRC collision |
Stream remains secure |
21 |
SRTP with multiple streams |
Encrypt multiple RTP streams |
All streams are secure |
22 |
SRTP with simulcast |
Use simulcast with SRTP |
All simulcast streams are encrypted |
23 |
SRTP with layered encoding |
Use SVC with SRTP |
All layers are encrypted |
24 |
SRTP with NAT traversal |
Use STUN/TURN with SRTP |
Stream traverses NAT securely |
25 |
SRTP with firewall |
Send SRTP through firewall |
Stream is transmitted securely |
26 |
SRTP with VPN |
Send SRTP over VPN |
Stream is encrypted end-to-end |
27 |
SRTP with IPv4 |
Use SRTP over IPv4 |
Stream is encrypted |
28 |
SRTP with IPv6 |
Use SRTP over IPv6 |
Stream is encrypted |
29 |
SRTP with transport switching |
Switch transport mid-session |
Encryption remains intact |
30 |
SRTP with fallback transport |
Fallback to alternate transport |
Stream remains secure |
31 |
SRTP with congestion |
Simulate network congestion |
Stream remains secure |
32 |
SRTP with bandwidth variation |
Vary bandwidth |
Stream adapts securely |
33 |
SRTP with authentication failure |
Tamper with packet |
Packet is rejected |
34 |
SRTP with integrity check |
Validate packet integrity |
Packet is verified |
35 |
SRTP with encryption overhead |
Measure encryption impact |
Acceptable latency observed |
36 |
SRTP with RTP mixer |
Use SRTP with mixer |
Mixed stream is encrypted |
37 |
SRTP with RTP relay |
Use SRTP with relay |
Relayed stream is encrypted |
38 |
SRTP with SFU |
Use SRTP with SFU |
Forwarded streams are encrypted |
39 |
SRTP with transcoder |
Transcode encrypted stream |
Stream is decrypted and re-encrypted |
40 |
SRTP with media gateway |
Use SRTP with gateway |
Stream is encrypted end-to-end |
41 |
SRTP with session renegotiation |
Renegotiate session |
Encryption is re-established |
42 |
SRTP with session timeout |
Simulate timeout |
Session ends securely |
43 |
SRTP with session resume |
Resume session |
Encryption resumes correctly |
44 |
SRTP with key lifetime expiry |
Expire key lifetime |
New key is negotiated |
45 |
SRTP with invalid cipher suite |
Use unsupported cipher |
Session fails to start |
46 |
SRTP with secure SDP |
Use SDP with crypto attributes |
Keys are exchanged securely |
47 |
SRTP with secure signaling |
Use secure signaling (TLS) |
Keys are protected |
48 |
SRTP with secure media negotiation |
Negotiate media securely |
SRTP is established |
49 |
SRTP with secure logging |
Log encrypted stream activity |
Logs do not expose content |
50 |
SRTP with compliance check |
Verify against RFC 3711 |
Implementation is compliant |
Jitter Compensation - Testcases
S.No |
Test Case |
Description |
Expected Result |
---|---|---|---|
1 |
Validate jitter buffer initialization |
Start RTP session |
Jitter buffer is initialized |
2 |
Validate jitter buffer with no jitter |
Send RTP with uniform delay |
Playback is smooth |
3 |
Validate jitter buffer with low jitter |
Introduce 10ms jitter |
Playback remains smooth |
4 |
Validate jitter buffer with high jitter |
Introduce 100ms jitter |
Playback is adjusted |
5 |
Validate jitter buffer with burst jitter |
Send bursty RTP packets |
Buffer compensates delay |
6 |
Validate jitter buffer under packet reordering |
Reorder RTP packets |
Playback order is corrected |
7 |
Validate jitter buffer with packet loss |
Drop random packets |
Playback continues with minimal disruption |
8 |
Validate jitter buffer with delay spike |
Introduce sudden delay |
Buffer absorbs spike |
9 |
Validate jitter buffer with clock drift |
Simulate clock skew |
Buffer adjusts timing |
10 |
Validate jitter buffer with variable network delay |
Vary delay dynamically |
Buffer adapts in real-time |
11 |
Validate jitter buffer with RTP over UDP |
Send RTP via UDP |
Buffer compensates jitter |
12 |
Validate jitter buffer with RTP over TCP |
Send RTP via TCP |
Buffer handles delay variation |
13 |
Validate jitter buffer with RTP over Wi-Fi |
Use Wi-Fi network |
Buffer smooths playback |
14 |
Validate jitter buffer with RTP over mobile |
Use mobile data |
Buffer handles jitter spikes |
15 |
Validate jitter buffer with RTP over VPN |
Use VPN tunnel |
Buffer compensates added delay |
16 |
Validate jitter buffer with RTP over satellite |
Use satellite link |
Buffer handles high jitter |
17 |
Validate jitter buffer with RTP over LAN |
Use wired LAN |
Minimal buffer adjustment needed |
18 |
Validate jitter buffer with RTP over congested network |
Simulate congestion |
Buffer compensates delay |
19 |
Validate jitter buffer with RTP pause/resume |
Pause and resume stream |
Buffer resumes smoothly |
20 |
Validate jitter buffer with RTP mute/unmute |
Mute and unmute stream |
Buffer handles silence periods |
21 |
Validate jitter buffer with RTP stream redirection |
Redirect stream |
Buffer reinitializes correctly |
22 |
Validate jitter buffer with RTP stream duplication |
Duplicate packets |
Buffer discards duplicates |
23 |
Validate jitter buffer with RTP stream switching |
Switch between sources |
Buffer adapts to new stream |
24 |
Validate jitter buffer with RTP stream mixing |
Mix multiple streams |
Buffer handles mixed timing |
25 |
Validate jitter buffer with RTP stream splitting |
Split stream to clients |
Each buffer compensates independently |
26 |
Validate jitter buffer with RTP stream forwarding |
Forward via SFU |
Buffer compensates forwarded jitter |
27 |
Validate jitter buffer with RTP stream transcoding |
Transcode mid-stream |
Buffer adapts to new timing |
28 |
Validate jitter buffer with RTP stream encryption |
Encrypt RTP |
Buffer handles encrypted timing |
29 |
Validate jitter buffer with RTP stream decryption |
Decrypt RTP |
Buffer compensates post-decryption |
30 |
Validate jitter buffer with RTP stream recording |
Record jittered stream |
Playback is smooth |
31 |
Validate jitter buffer with RTP stream playback |
Play recorded stream |
Buffer compensates recorded jitter |
32 |
Validate jitter buffer with RTP stream under load |
Simulate CPU load |
Buffer maintains playback |
33 |
Validate jitter buffer with RTP stream under memory pressure |
Simulate low memory |
Buffer adapts size |
34 |
Validate jitter buffer with RTP stream under power saving |
Enable power saving |
Buffer maintains timing |
35 |
Validate jitter buffer with RTP stream in browser |
Use WebRTC |
Buffer compensates jitter |
36 |
Validate jitter buffer with RTP stream in native app |
Use native client |
Buffer compensates jitter |
37 |
Validate jitter buffer with RTP stream in embedded device |
Use IoT device |
Buffer handles jitter |
38 |
Validate jitter buffer with RTP stream in conferencing |
Use in video call |
Buffer ensures smooth playback |
39 |
Validate jitter buffer with RTP stream in broadcasting |
Use in live stream |
Buffer compensates delay |
40 |
Validate jitter buffer with RTP stream in surveillance |
Use in CCTV |
Buffer ensures real-time playback |
41 |
Validate jitter buffer with RTP stream in gaming |
Use in game stream |
Buffer minimizes latency |
42 |
Validate jitter buffer with RTP stream in education |
Use in e-learning |
Buffer ensures clarity |
43 |
Validate jitter buffer with RTP stream in healthcare |
Use in telemedicine |
Buffer ensures reliability |
44 |
Validate jitter buffer with RTP stream in finance |
Use in trading platform |
Buffer ensures low-latency playback |
45 |
Validate jitter buffer with RTP stream in remote control |
Use in drone control |
Buffer minimizes delay |
46 |
Validate jitter buffer with RTP stream in AR/VR |
Use in immersive media |
Buffer ensures sync |
47 |
Validate jitter buffer with RTP stream in multi-language audio |
Use multiple audio tracks |
Buffer syncs all tracks |
48 |
Validate jitter buffer with RTP stream in multi-camera setup |
Use multiple video feeds |
Buffer syncs feeds |
49 |
Validate jitter buffer with RTP stream in hybrid cloud |
Use cloud-edge setup |
Buffer compensates network variance |
50 |
Validate jitter buffer with RTP stream in 5G network |
Use 5G transport |
Buffer handles ultra-low latency |
Reference links