I want to send a file .mp3 in RTP packets to a client. What I know is, how many bytes I get from .mp3 file to send in each RTP packet, that is, how much bytes of the .mp3 file I place in the payload field of each RTP packet?
Thanks for the help.
Greetings!
You should look at RFC 5219 - A More Loss-Tolerant RTP Payload Format for MP3 Audio.
It's not as simple as "how many bytes...". This also references the older RFCs for MP3.
Related
Can SIPp support play audio and video file together in a call? Thanks.
I am able to initiate a sip audio call. What I want to achieve is, after the callee picks the call, play a audio and video file together.
I want to achieve this using the command line arguments. Has someone done this before?
You should be able to do it the same way you are playing a pre-recorded audio file. Please see the relevant section in the SIPp document:
PCAP (media) commands
PCAP play commands (specified using play_pcap_audio / play_pcap_video attributes) allow you to send a pre-recorded RTP stream using the pcap library.
Choose play_pcap_audio to send the pre-recorded RTP stream using the "m=audio" SIP/SDP line port as a base for the replay.
Choose play_pcap_video to send the pre-recorded RTP stream using the "m=video" SIP/SDP line port as a base.
The play_pcap_audio/video command has the following format: play_pcap_audio="[file_to_play]" with:
I have mp4 stream available over HTTP to single client
How to broadcast it to multiple clients?
I can write code that will read MP4 stream over HTTP and copy to multiple clients.
But is there some initial frame, header in MP4 stream?
May be some framework or existing server can broadcast mp4 stream?
Use VLC, it has nice streaming possibilities (VLC is for Video Lan Client, by the way).
Just fire up VLC, go to File->Stream..., select the source you want to distribute, choose e.g. http as method, transcode (or don't), and select the port to listen on. From there on, your clients can connect :)
I'm trying to save 5 seconds .mov segments of an RTSP stream with VLC. First I tried openRTSP and ffmpeg but both of them gives incorrect output (Index missing etc). I've read a lot of the VLC cli, but havn't had any luck of saving an RTSP stream as segments.
If I use the VLC GUI I can both save segments as saving snapshots (PNGs) but I need to do this via CLI.
mov files are not streamable. [they are right in saying index file missing]. I don't even know how you are sending them over rtsp there is no rtp payloader i am aware of for mov/mp4 format.
I have a working POST request that will send a CAF audio file (recorded on the iPhone with AVAudioRecorder) to a web server -- the request itself is structured similarly to Send an audio FILE and JSON string in iOS.
The server correctly receives the file, but I'm having trouble converting it into a format that will play directly in a browser. In addition to every audio/iPhone/server post I could find on stack overflow, I've looked into ffmpeg and HTML5 audio as well, but couldn't find clear instructions on how to convert the received audio binary to a browser-playable format. Essentially, I need to understand how to accomplish Arun's server-side suggestion here.
I know that Audacity, Soundbooth, etc. will allow you to use the raw data and save it as another format, but I need this to be done programmatically on the server. If there are any suggestions they would be very much appreciated! Thank you.
I'm using some code related to RTSP, RTP to listen to various RTSP Streams Using FFMPEG, it works!
BUT
The noise is being decoded in such a way that every 10 seconds a glitch in the ASF decoding of the stream occurs, where the Volume Peaks and makes a loud Popping sound.
Generally the sound you hear when a packet is corrupted...
I'm just wondering if anyone can help me with where to look for Troubleshooting, when working with WMA ASF Audio Streams.
Any help/tips/pointers are appreciated.
I'm not sure if it's in the RTSP Parser, Data Buffer, WMA Decoder...
I know nothing about WMA/ASF, but have you checked the sequence numbers in the RTP headers are contiguous (how you would find out what these are will depend on what RTSP/RTP library you are using)? At least then you will know whether packets are going missing or not.