I would like to develop a very tiny and small RTSP client to get the video stream from network cameras. Does anybody know where can I find a simple explanation of the protocol and some good examples?
Best regards,
You connect to the camera via RTSP protocol to query its capabilities, identify streams and prepare/start transmission.
RFC 2326 - Real Time Streaming Protocol (RTSP)
As a part of initialization and handshaking, you will discover available streams.
RFC 4566 - SDP Session Description Protocol
Then you will set up RTP session(s) to receive data, over UDP or sharing the same TCP connection.
RFC 3550 - RTP A Transport Protocol for Real-Time Applications
RFC 4571 - Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection-Oriented Trans
To decode media streams you will convert the payload into pure data you need for further processing. With IP cameras your primary interest is perhaps MPEG-4 AVC (H.264):
RFC 3984 - RTP Payload Format for H.264 Video
RFC 6184 - RTP Payload Format for H.264 Video
This looks like some (introductory) reading.
Try GStreammer library. It is modular, wery flexible library, which can be used for streamming (both client and server). Just check the docs and pick right plugins.
GStreammer could be used in two ways: as a commandline tool or as a library in your project, depending on your requirements.
Related
I'm trying to build RTP media projector for only audio streams.
A user will create a session with the server and possibly broadcast audio stream.
The server will send to the user audio streams of all the other active users.
Can the server send media from a single port or does it need to be able to use a range of ports for sending? (I know it needs to listen to ports 1024 - 65535).
Does the server need to use ICE or can it just respond to the SDP request right away?
How does RTCP works in this scenario? Does the server sends quality of service feedback in the name of clients or acts as a client and sends feedback for himself?
What does the server do with quality of service feedback from the clients?
Does the server need to do something with the media packets like changing timestamps or just deliver them as is, assuming all clients are using the G.711 codec?
Thanks
The MCU can use a single port if it is the passive side in the peer-to-peer connection or a separate port for each session if it is the active side.
The MCU can act as a Translator and just forward RTCP packets from clients but this might result with high bandwidth usage. A more complicated MCU can parse RTCP packets and generate RTCP reports from this info.
The MCU need to decrypt and re-encrypt the RTP packets but as long as all the participants are using the same codec, there is no need for transcoding.
The info can be found in the RTP rfc
http://www.ietf.org/rfc/rfc3550.txt
I got a pcap file with voip conversation, how i can separate RTP packets from the other packets?
I can obtain sip packets, but I can't differenciate the RTP packets from the rest.
Search for RTP headers as defined in RFC3550 within your file. Or better use pcap-filter, for instance with this wiki (look for "Q: What is a good filter for just capturing SIP and RTP packets?").
If you want to see the RTP traffic in wireshark then:
Select Analyze->Display Filters...
Select "UDP", OK
Right click on any UDP packet and select "Decode as..."
Select "RTP" from the list, OK
Now you can see all RTP packets.
Hope that helps. :)
p.s. edited to note that this is for Wireshark. Thanks to a commentor for pointing that out!
Check #macs recommendation about PCap filter. If this cannot satisfy your needs (e.g. you need to filter out RTP packets of specific SIP session) there's no simple way. You need to parse SIP messages, retrieve RTP port numbers, takes packets going to/from these ports in particular time period and (optionally) check if these packets are RTP by checking their headers (magic number in headers)
An open source software that extract the RTP/RTCP packets from a pcap file are:
rtpbreak: http://dallachiesa.com/code/rtpbreak/doc/rtpbreak_en.html
xplico: http://www.xplico.org
From the source code you can view and understand the methodologies used.
I can obtain sip packets, but I can't differenciate the RTP packets
from the rest.
If you are able to decode the SIP, then you can find (inside INVITE message) the SDP message. If you decode it you can find the IP and PORT of RTP "stream" (and RTCP => port + 1). With these informations you can identify uniquely the RTP and RTCP packets. Keep in mind that there are often packages (with the same IP-PORT) with the STUN protocol which must be separate from RTP.
You have to consider where is the packet capture (network context and constraints), you may take into account NAT.
I am developing a client program which will display the media captured from IP camera. So I want to whether the RTP packets using RTSP protocol contain both audio and video if contains both how should I extract it?
RTSP stream does not carry video/audio itself, it provides a method to control independent RTP video and audio streams (they are in turn independent one from another).
One of the options though is when RTP streams are tunnelled through RTSP connection, in which case all communication might be taking place through single TCP connection.
You can read the SDP returned in the SETUP request to the RtspServer.
There should be a MediaInformation for each stream available.
That will tell you if there is audio or video etc...
http://en.wikipedia.org/wiki/Session_Description_Protocol
How do I write a C++/MFC program to make a server as a bridge for clients to stream their audio? I have been told to use UDP and RTP protocol but due to my lack knowledge of media streaming, I couldn't make it work. What is relationship between UDP and RTP and steps needed for server to listen, accept and handle packet transfer between client to client.
As unwind said, generally RTP runs on top of UDP. It's called a conectionless protocol.
This is the specification of UDP: http://www.ietf.org/rfc/rfc768.txt
An this is the specification of RTP: http://www.ietf.org/rfc/rfc1889.txt
You can find very useful information about RTP on this site. There are different libraries and docs.
It's possible to write a "RTP forwarder" application.
RTP generally runs on top of UDP, to get away from TCP's streaming behavior, TCP always delivers data in-order, which is not optimal for real-time applications.
It might be possible to do a "dumb" forwarder that is not RTP-aware, but instead is configured to e.g. accept UDP packets to port X, and forward all traffic to host:Y, packet by packet. Not sure if that works in practice, though.
I'm developing the application that sending MPEG video over IP network using RTP protocol.
In order to test it I'm looking for a software tool that can measure network jitter, recognize RTP packets reordering events and show results as a graph.
Any help is highly appreciated.
You can do a trace and then analyze it in Wireshark. Probably it won't recognize the RTP streams at first (it doesn't know the port number), but you can select one packet and choose 'Decode As.. RTP'. After that, go to 'Telephony / RTP / Stream Analysis' and you will get comprehensive RTP statistics.
The solution is to use Wireshark: Open Main menu->telephony->RTP->Stream analysis...
More information is here: http://wiki.wireshark.org/RTP_statistics
#Dima, if you are looking for a software tool capable of measuring network jitter, I suggest you check out http://www.netrounds.com. This is a cloud-controlled software solution for IPTV MPEG transport monitoring in line with ETR 101 290. You can measure both PCR and RTP jitter.
RTP jitter will show any network jitter problems in your transport path, whereas PCR jitter will also include any jitter introduced by encoding/transcoding.
There is a free version of Netrounds available that supports monitoring of two IPTV MPEG transport streams (channels) concurrently. Channels are received by the probe/agent by IGMP join messages sent to the corresponding multicast groups.
DISCLAIMER: I am affiliated with Netrounds.