I have been trying to stream via RTP and RTSP using vlc. The strange thing is that when streaming is done via RTSP (unicast), RTCP generates both Sender Reports and Receiver Reports, but when streaming is done via RTP multicast only Sender Reports are generated. The clients are not generating any Receiver Reports. I use wireshark for packet sniffing.
Any help will be appreciated.
Related
I want to develop an android app which uses push to talk or Voip to communicate between 2 mobile devices using a Wi-Fi connection without using a data or the internet.
Is it possible to develop this?
I am looking for a 'free' Wi-Fi Walkie Talkie but I don't know how to start doing it?
For a very simple, initial version, I'd do the following:
Assign a static IP address on each phone
Record the audio and packetise it into a UDP stream that you'll send to the remote IP address
Repackage again the UDP stream and play the audio.
Microphone ---> Samples ----> UDP (over WiFi) ----> Samples ----> Speaker.
There are plenty of enhancements that you can add over the time:
Device discovery
Signalling separate (SIP or custom messages via TCP or UDP to indicate when to start the audio transmission and negotiate the media channel that will be used).
Support for third party devices (SIP)
Use standard paketization using RTP and encode the audio using G711alaw/G711ulaw...
There are plenty options, but you should start by capturing the audio, and sending it over the network, even make packets of 2 seconds with the incurring delay, but that's a point to start. Then, you should lower the packet length to contain 20ms of audio to avoid delays in the transmission.
Hope this helps.
I need to build a RTMP parser . For that I need to know from where the RTMP packet starts and where it ends in a bytestream. I analysed the RTMP stream using wireshark.What I observed is that RTMP packet is packaged(encapsulated) inside TCP packet which is in turn packaged(encapsulated) inside IPV4 packet. This was the behavior for all the RTMP packets except Video data RTMP packets and Aggregate RTMP packets.To build a parser I need to know from where the RTMP packet is starting and where it is ending.But with Video data RTMP packet and Aggregate RTMP packets its not getting possible because these packets are not packaged inside IPV4 packet.The data for RTMP packet is shown in a different window in wireshark(named Unchunked RTMP).Can anyone help me know how wireshark is getting to know where the TCP packet is ending and where the RTMP Video packet and RTMP Aggregate packet is starting ??? What is the logic of Wireshark for this or what I can do to achieve the same ???
We have implemented RTSP server on our MCU. For testing purpose we are using VLC media player as a client. We coded our MCU such a way that only after receiving PLAY command from client, MCU reads data from camera. And we are seeing MCU receives data from camera and streams thorugh RTSP. We could see data streaming from server through UDP on Wireshark. And Also Codec information getting dispalyed on VLC media player. But video doesn't get played in VLC.What could be the issue?
Below is our SDP file info
"v=0\r\ns=Unnamed\r\ni=N/A\r\nc=IN IP4 sumukha-PC\r\nt=0 0\r\na=tool:vlc 2.2.2\r\na=recvonly\r\na=type:broadcast\r\na=charset:UTF-8\r\na=control:rtsp://192.168.1.100:8555\r\nm=video 0 RTP/AVP 96\r\nb=RR:0\r\na=rtpmap:96 H264/90000\r\na=fmtp:96 packetization-mode=1\r\na=control:rtsp://192.168.1.100:8555/trackID=0\r\n\r\n");
Thanks,
Ck
The error message tells you what is going wrong:
live555 error: no data received in 10s, aborting
The usual reasons for this are firewalls or NAT?
You can select the RTP over RTSP (TCP) option in the VLC preferences to verify this. If the stream works over TCP, UDP packets are being blocked somewhere.
I suggest using ffmpeg and ffplay to test your streaming from a RTSP source.
It is command line, but the information and logs are very helpful
I had an confusion of which I searched but didn't get any answer.
I am streaming a video file using RTSP over HTTP tunneling then when I see the wire-shark to see the packets source and destination and protocol I saw a UDP protocol although i am using RTSP over HTTP tunneling.
Can any one tell me why don't I see the TCP protocol instead of UDP protocol for packets being sent from source to destination?
RTSP sets up RTP streams for streaming the video and audio. They are probably being sent over UDP instead of tunneling over the existing TCP connection.
Just because RTSP is being tunneled over HTTP doesn't guarantee that the RTP streams will be configured to use the same socket. It's up to the client to request the transport, but you can change the server to only support the interleaved transport. See this answer https://stackoverflow.com/a/3536969/759140
RTSP is a network control protocol. According to Wikipedia,
The default port for the RTSP protocol is 554 for both UDP (deprecated
and very rarely used) and TCP transports.
Conclusion: UDP or TCP will be used, depends on which server you use and settings.
Very true but if you specify to use tcp-transport in ffmpeg it should use TCP if the server is configured for it.
If you don't specify any transport option, new versions of ffmpeg will try UDP, if that fails it will retry the connect or TCP.
I am trying to setup a network where I am reading a broadcast with live555 and then trying to send out an RTSP server I was reading the tutorials on the website and couldnt find what I was doing wrong. My process was I opened up a streaming network through VLC to stream udp to the port that live555 was listening to then I opened testOnDemandRTSPServer for live555 and then opened a network stream from another VLC to listen to the rtsp stream but was getting no output does anyone know what I did wrong
I found out the problem was that I was using a virtual machine and had bad network setting and had to disable something, I also changed the code in testOnDemandRTSPServer.cpp to line 340 and changed inputStreamIsRawUDP to True