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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WebRTC: SDP in Firefox with VP9 encoding

I am not able to connect a call from Firefox to Firefox using VP9, allthough I have tried modifying the SDP in several different ways. I have a site similar to https://webrtc.github.io/samples/src/content/peerconnection/munge-sdp/, where I simply…
anters
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Why does the m= line in sdp contain a port? Isn't this redundant with candidates?

According to RFC 4566 https://www.rfc-editor.org/rfc/rfc4566#page-22, the m= line in an sdp contains a port: m=audio 49170 RTP/AVP 0 Aren't ports determined by ICE candidates already? These contain ports as well: a=candidate:1 1 UDP 2130706431…
Oliver Hausler
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Is it possible to combine media tracks/streams in WebRtc

I have two webcams that I would like to share, but I do not want to have to negotiate two sets of the signaling process (SDP, ICE, all that jazz). Is there any way for me to combine these two stream into one before it goes to another user, or I am…
user1634494
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Trouble connecting android bluetooth client socket to ubuntu server socket

I am writing an Android app that should exchange data to a server over Bluetooth, with the server side on a PC running Ubuntu, using the bluez library, in C (or C++). My Android app fails (IOException) when I try to connect to my server socket on…
UlrikK
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How to exchange streams from two peerconnections with offer/answer

I am trying to setup video chat where two peer connections exchange video. This happens after creating a data channel. So this is the process of events: offerer creates data channel, offerer creates and offer, answerer creates an answer. So far so…
Fab
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What is the minimal SDP answer to receive WebRTC Audio and Video?

I am currently trying to stream WebRTC MediaStreams to my Server where it will be recorded. Sadly there is no native Java webRTC endpoint, so I want to implement this special case myself. Now, given an offer sdp and my server's public IP, how do I…
Joel Pepper
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Error : Failed to push down transport description: Failed to apply remote fingerprint

I wrote two unit tests to verify whether the offer/answer mechanism is correctly implemented in my application. The first unit test verifies the behaviour of a peer when an offer is received, whereas the second unit test verifies the behaviour of a…
Giuseppe Pes
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Modifying SDP constraints for WebRTC in FireFox

I am trying to reduce the bitrate in a RTCPeerConnection within FireFox. I have successfully been able to do within Chrome. I am modifying the SDP string that is automatically generated by FireFox after calling the createOffer method. My callback…
Benjamin Trent
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Cannot connect to Bluetooth/BlueZ server: Could not create RFCOMM socket

I am trying to create a simple and custom Bluetooth connection between my Linux laptop and Android phone. On the Linux side of things, I use the BlueZ libbluetooth library to set up a server. The code for it is very similar to others I've seen…
gnychis
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WebRTC: what happens if both peers send an offer?

Usually, one of the peers creates an offer while the other one answers it. What happens if both peers try to set up connections by creating an offer at the same time?
knshn
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Using ZeroMQ on Infiniband SDP

I am trying to run the ZeroMQ's local_thr/remote_thr on SDP (infiniband) compiled on MSVS2012. But it's not connecting. On IPoIB it is working properly. Operating System is Windows Server 2008 R2. On further investigation I found out that select()…
Eiraj
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Grant the root privilege to the application

I have an application which consists of the native part and higher layer. Therefore, it needs to be granted the root privilege to run some native functions. As the article I post before, I added a piece of code taken from Bluez Utils(hidd.c). if…
Sam
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Use sdp session to play stream?

I have a sdp session structure which contains all the relevant information like audio port, video port etc. Now, how can this sdp session structure be used to play the media using libvlc library ? Pls help. Thanks in advance.
parvez
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how to send mu-law (G.711) codec wav file on a network SIP phone and play there

I am struggling to send a mu-law (G.711) codec wav file as binary data over a network on a SIP phone and want to play this wav file on that SIP phone. Below is the code in C#. Can anybody tell me what I am doing wrong? private void…
vicky
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In need of GSM codec specifications in specific format

What is the default encoded frame size of gsm codec? What is the default decoded frame size? What are the media attreibutes? What is the codec id? For example same questions for G729's answers are 20, 160, "fmtp:18 annexb=no" and 18 respectively.…
S. M. Shahinul Islam
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