Questions tagged [sdp]

The Session Description Protocol (SDP) describes multimedia sessions for the purpose of session announcement, session invitation and other forms of multimedia session initiation.

Session directories assist the advertisement of conference sessions and communicate the relevant conference setup information to prospective participants. SDP is designed to convey such information to recipients. SDP is purely a format for session description - it does not incorporate a transport protocol, and is intended to use different transport protocols as appropriate including the Session Announcement Protocol (SAP) , Session Initiation Protocol (SIP) , Real-Time Streaming Protocol (RTSP) , electronic mail using the MIME extensions, and the Hypertext Transport Protocol (HTTP) .

Protocol Structure - SDP (Session Description Protocol)

SDP messages are text messages using the ISO 10646 character set in UTF-8 encoding. SDP Session description (optional fields has an *) is:

v= (protocol version)

o= (owner/creator and session identifier).

s= (session name)

i=* (session information)

u=* (URI of description)

e=* (email address)

p=* (phone number)

c=* (connection information - not required if included in all media)

b=* (bandwidth information)

One or more time descriptions (see below)

z=* (time zone adjustments)

k=* (encryption key)

a=* (zero or more session attribute lines)

Zero or more media descriptions (see below)

Time description

t= (time the session is active)

r=* (zero or more repeat times)

Media description

m= (media name and transport address)

i=* (media title)

c=* (connection information - optional if included at session-level)

b=* (bandwidth information)

k=* (encryption key)

a=* (zero or more media attribute lines)

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How can I change the default Codec used in WebRTC?

I've been googling a way to change codec in Chrome's implementation of WebRTC, but there doesn't seem to be a way. How can I change the default codec used(audio or video) in a WebRTCpeer connection in Chrome?
Tony
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WebRTC: Use same SDP for multiple peer connections?

Is it possible to use same SDP in multiple peer connections? I'm building video conference using WebRTC. The idea is that caller, using some signaling mechanism, send broadcast message to all other users with it's SDP (same SDP for each user) and…
SET001
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Minimum SDP for making a H264 RTP stream?

I'm looking for an example of a minimum necessary SDP for setting up a H264 video stream.| The assumption is that the receiver can play H264 as long as it gets the required parameters through SDP. I have found a related document herehowever it uses…
TheMeaningfulEngineer
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WebRTC multiple video streams in same peerConnection

I am working on a WebRTC client and I would like to allow the clients to modify the ongoing audio/video session to add or remove an audio or video stream. So for instance if there is an ongoing audio/video call between two clients, one client can…
dirtandfilth
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How to decode sprop-parameter-sets in a H264 SDP?

What is the meaning of Base64 decoded bytes in sprop-parameter-sets in SDP for a h264 stream? How can I know the video size from this example? SDP example: sprop-parameter-sets=Z0IAKeNQFAe2AtwEBAaQeJEV,aM48gA== First part decoded from Base64 to…
Cipi
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How to generate SDP file from FFMPEG

So, I have been working with FFMPEG on a project that involves streaming video from one computer to another across the internet with RTP. I want to take that into ffmpeg and use ffserver to display it on a local network. As I understand it, you need…
MobilityLab
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How to set remote description for a WebRTC caller in Chrome without errors?

I hope there is no flaw in the logic. Step 1: caller creates offer Step 2: caller sets localDescription Step 3: caller sends the description to the callee //------------------------------------------------------// Step 4: callee receives the offer…
Satnam Sandhu
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WebRTC one-way video call

we are fiddling around with WebRTC in our company. And i stumbled upon a weird thing, which i'm not sure is by design in WebRTC or an implementaiton error on our side. We currently have a simple WebApp which displays a Contact-List of online…
Sven Eppler
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Force TCP for WebRTC PeerConnections

Is it possible to force a TCP tunneled (TLS) connection with WebRTC? We are developing a WebRTC application for our business, but we are experiencing some major issues with incoming UDP streams caused by our internal network. We are already using a…
Rolf Beh
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How to start RTP stream inside a SIP/SDP call

I've managed to set up a SIP call using the JAIN-SIP API for Java. Now I would like to stream some video once a dialog has been established. I've read that this is possible with SDP and RTP, and I've found multiple examples on how to define a…
user50685
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How to use webrtcbin create offer,only receive video

the gstreamer webrtc demo works fine.but all demo has a small problem: all webrtcbin that created offer must have some video/audio data to send. i want use webrtcbin create offer,and only receive video data from other webrtc peer. all demo pipeline…
宝蓝娃娃
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How to tell if pc.onnegotiationneeded was fired because stream has been removed?

I am dealing with a RTCPeerConnection (pc) which has an event handler named onnegotiationneeded. The "onnegotiationneeded" is triggered when a complete media stream is added or removed with pc.addStream or pc.removeStream. Is it possible to tell…
Benny Code
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Identify h264 profile and level from profile-level-id in sdp?

In a SIP video call, the receiver of the video stream respond with the capabilities of its decoder. The parameter which defines that is the profile-level-id. Here is an example value of the profile-level-id parameter: 428014 RFC 6184 defined that…
TheMeaningfulEngineer
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Is sprop-parameter-sets or profile-level-id the SDP parameter required to decode H264?

I'm trying to understand what is the required parameter in SDP to be able to decode H264 from RTP packets. This is an related to this question, for the answer to that one works only in small number of cases. Example I am streaming from VLC with the…
TheMeaningfulEngineer
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Stream RTP to FFMPEG using SDP

I get RTP stream from WebRTC server (I used mediasoup) using node.js and I get the decrypted RTP packets raw data from the stream. I want to forward this RTP data to ffmpeg and from there I can save it to file, or push it as RTMP stream to other…
Johnathan Kanarek
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