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 to stream video in android device via .sdp file from android device

I had anapplication to stream video(without audio) from android device(Blaze board) to PC. For that, I'm using vlc player to view streamed video by using the following command in command line vlc stream.sdp This .sdp file would be generated from my…
Dhasneem
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Same media format for audio and video on RTSP

Our company develops a camera surveillance software and we mainly use RTSP for communication with the devices (But we support any protocol required) and we have developed our own RTSP Client and parsers Today we were working on an integration of a…
Eric
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sdp: The order of m-lines in subsequent offer doesn't match order from previous offer/answer

**i am using React Native and getting these two errors in my console log : Failed to set local offer sdp: The order of m-lines in subsequent offer doesn't match order from previous offer/answer. Failed to set remote answer sdp: Called in wrong…
user19434921
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How do I rescue my WebRTC connection when my SDP offer is swallowed?

I'm using AWS Kinesis Video Signalling Channels to set up a WebRTC connection, with help from the Amazon Kinesis Video Streams WebRTC SDK for JavaScript (on whose repo I have cross-posted this question:…
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How to use the constant resolution during webrtc video transmission?

I am using janus to build my webrtc SFU server. I need the chrome browser to send the video resolution from a start to a fixed value and remain unchanged during the transfer. Where should I set it? I tried setting the degradationPreference in the js…
chunqiulfq
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Lollipop set remote description error : rtcp-mux must be enabled when BUNDLE is enabled

I tried to make Android WebRTC app I tested on Android Oreo and Pie, the app works well. But on Android Lollipop and Marshmallow, throw a error when I call setRemoteDescription function. (org.webrtc.SdpObserver::onSetFailure(String)) Failed to set…
Jay Son
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Missing packets error when streaming rtsp to rtmp using ffmpeg

I'm using ffmpeg to do RTSP to RTMP streaming, the input is an sdp file describing one video stream and one audio stream, when I test the RTSP using ffplay,it works fine ffplay -protocol_whitelist "file,udp,rtp" -strict -2 -i media.sdp but when I…
ArL
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ICE connection fails in WebRTC

We are trying to connect a browser (client) with the aiortc library (server, which sends a single video stream). Currently, the connection is successfully established (on signalingstatechange is stable). However, the media connection is never…
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WebRTC setRemoteDescription error

I wanted to test connection between my c++ app and browser javascript. For now i have no server involved and i'm doing signaling stuff on my own by standard IO. The issue is when I try to pass sdp generated by native app to javascript to create…
Fake
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Native WebRTC dropping frames

Summary: How do I stream high quality video using WebRTC native? I have an h264 stream that's 1920x1080 at about 30fps. I can currently stream this from a server on localhost to a native client on localhost just fine. I wrote a WebRTC server using…
user2690730
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Using SDP media type application with RTP/AVP (m=application RTP/AVP )

I am trying to get familiar with the anatomy of a SIP SDP. Here is a sample SDP from my Tandberg VC unit. v=0 o=tandberg 1 3 IN IP4 192.168.1.94 s=- c=IN IP4 192.168.1.94 b=AS:768 t=0 0 m=audio 47032 RTP/AVP 97 98 99 100 101 9 15 8 0…
Tezro Solutions
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"iceConnectionState" is "failed" without any error

Browser: Chrome 62.0.3202.94, (64 bit) OS: Ubuntu 16.04 I'm trying to create a data connection. I've done this before with this code a couple of months ago, but now I can't quite get it to work. The problem is that when I call…
manidos
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GStreamer Launch RTSP Server for ReStreaming IP Camera H264

I am going to use multiple clients on different computers to be able to view video of an IP Camera stream url. Because the Ip camera has limitations on the number of connected clients, I want to setup a streamer for this purpose. I googled and tried…
M.Mahdipour
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Is the turnserver not being detected?

I have set up my own turnserver to prevent cross-domain errors from occuring. When the client needs to use a TURN server rather than STUN there is no video feed. Messages are getting through however. My main.js (a takeoff from WebRTC…
Brett Grentell
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How to use WebRTC without an answer?

In the absence of a signalling server for coordinating the initial exchange, does WebRTC provide any way to allow the responder to send information freely to the caller, if the responder has only received an offer and has no other methods of…
Shien
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