A parser for SDP partially compliant with RFC 4566. More about Sesssion Description Protocol. Another interesting RFC is the RFC draft for Opus codec. This parser was written to be used in conjuction with WebRTC. This way you can easily modify settings like codec parameters.
Using
- npm:
npm install sdpparser. See below an example usingrequire. - directly: download the archive from releases and include
SdpParser.jsandjscommon.js
This can be used either from a browser or on the server side.
var SdpParser = require("SdpParser");
var filterSdp = function(sdpText) {
var sdp = SdpParser.parse(sdpText);
// ... do your changes to the sdp object ...
sdpText = SdpParser.format(sdp);
return sdpText;
}
peerConnection.createOffer(function(sessionDescription) {
sessionDescription.sdp = filterSdp(sessionDescription.sdp);
peerConnection.setLocalDescription(sessionDescription);
});
Given the following SDP
v=0
o=- 718035783275703419 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS 8pjbfuaudqpe5s3uA
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:Lw5NgNWQ
a=ice-pwd:qfy9wvi9J3g7G
a=fingerprint:sha-256 36:B6:83:67:03:BB:
a=setup:active
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3140675822 cname:STNh4RdVs
a=ssrc:3140675822 msid:8pjbfuaudq
a=ssrc:3140675822 mslabel:8pjbfua
a=ssrc:3140675822 label:40fff5e6-d618-4b7a
the parsed JSON object will be
{
"version": 0,
"origin": {
"username": "-",
"sessionId": "718035783275703419",
"sessionVersion": 2,
"networkType": "IN",
"addressType": "IP4",
"unicastAddress": "127.0.0.1"
},
"sessionName": "-",
"timing": {
"start":0,
"stop":0
},
"group": "BUNDLE audio video",
"msid-semantic": " WMS 8pjbfuaudqpe5s3uA",
"media": [
{
"type": "audio",
"port": 9,
"protocol": "UDP/TLS/RTP/SAVPF",
"payloads": [
{
"id":111,
"rtp": {
"codec": "opus",
"rate": 48000,
"codecParams": 2
},
"fmtp": {
"params": {
"minptime":10,
"useinbandfec":1
}
}
},
{
"id":103,
"rtp": {
"codec": "ISAC",
"rate":16000
}
},
{
"id":104,
"rtp": {
"codec": "ISAC",
"rate":32000
}
},
{
"id":9,
"rtp": {
"codec": "G722",
"rate":8000
}
},
{
"id":0,
"rtp": {
"codec": "PCMU",
"rate":8000
}
},
{
"id":8,
"rtp": {
"codec": "PCMA",
"rate":8000
}
},
{
"id":106,
"rtp": {
"codec": "CN",
"rate":32000
}
},
{
"id":105,
"rtp": {
"codec": "CN",
"rate":16000
}
},
{
"id":13,
"rtp": {
"codec": "CN",
"rate":8000
}
},
{
"id":126,
"rtp": {
"codec": "telephone-event",
"rate": 8000
}
}
],
"connection": {
"networkType": "IN",
"addressType": "IP4",
"connectionAddress": "0.0.0.0"
},
"rtcp": "9 IN IP4 0.0.0.0",
"ice-ufrag": "Lw5NgNWQ",
"ice-pwd": "qfy9wvi9J3g7G",
"fingerprint": "sha-256 36:B6:83:67:03:BB:",
"setup": "active",
"mid": "audio",
"extmap": [
"1 urn:ietf:params:rtp-hdrext:ssrc-audio-level",
"3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
],
"sendrecv": true,
"rtcp-mux":true,
"maximumPacketTime": 60,
"ssrc": [
"3140675822 cname:STNh4RdVs",
"3140675822 msid:8pjbfuaudq",
"3140675822 mslabel:8pjbfua",
"3140675822 label:40fff5e6-d618-4b7a"
]
}
]
}
Please note that
payloadshave been aggregated- for some SDP parameters, the object property names are expanded
- the properties with multiple values are aggregated into an array
If you want to involve in this project as a developer please read the short development guide.
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