Hello.
I have compiled the Mediasoup library with WebRTC adding logs to see the device’s capabilities and I have added the library to iOS project with this wrapper: https://github.com/ethand91/mediasoup-ios-client
I check the codecs that WebRTC supports from my app:
let defaultRTC = RTCDefaultVideoDecoderFactory()
let defaultEncodeRTC = RTCDefaultVideoEncoderFactory()
for codec in defaultRTC.supportedCodecs() {
print(codec.name)
print(codec.parameters)
}
for codec in defaultEncodeRTC.supportedCodecs() {
print(codec.name)
print(codec.parameters)
}
LOGS:
H264
["profile-level-id": "640c33", "level-asymmetry-allowed": "1", "packetization-mode": "1"]
H264
["profile-level-id": "42e033", "level-asymmetry-allowed": "1", "packetization-mode": "1"]
VP8
VP9
I get the capabilities from our server and it supports h264 but when I pass them to Mediasoup it doesn’t load h264.
I have consulted the offer created by Mediasoup from the library and it doesn’t include h264.
json Handler::GetNativeRtpCapabilities(const PeerConnection::Options* peerConnectionOptions)
{
MSC_TRACE();
std::unique_ptr<PeerConnection::PrivateListener> privateListener(
new PeerConnection::PrivateListener());
std::unique_ptr<PeerConnection> pc(
new PeerConnection(privateListener.get(), peerConnectionOptions));
(void)pc->AddTransceiver(cricket::MediaType::MEDIA_TYPE_AUDIO);
(void)pc->AddTransceiver(cricket::MediaType::MEDIA_TYPE_VIDEO);
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
// May throw.
auto offer = pc->CreateOffer(options);
auto sdpObject = sdptransform::parse(offer);
auto nativeRtpCapabilities = Sdp::Utils::extractRtpCapabilities(sdpObject);
MSC_DEBUG("GetNativeRtpCapabilities offer:\n%s", offer.dump(4).c_str());
return nativeRtpCapabilities;
}
LOGS
GetNativeRtpCapabilities offer:
v=0
o=- 3864646501880913761 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=msid-semantic: WMS
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:JuqU
a=ice-pwd:zlVuwktX7FdAJYbqRyP2QXCE
a=ice-options:trickle
a=fingerprint:sha-256 E8:82:F2:DC:B4:90:A9:EF:5D:F4:4B:8D:B5:51:CA:DB:EC:CD:93:A1:85:0A:20:80:89:35:43:72:DB:E9:19:25
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:- a1845c7b-cddd-4790-86ad-dfd5f65f3a31
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/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:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1051213426 cname:wWzqvgrPSbEVsUBN
a=ssrc:1051213426 msid:- a1845c7b-cddd-4790-86ad-dfd5f65f3a31
a=ssrc:1051213426 mslabel:-
a=ssrc:1051213426 label:a1845c7b-cddd-4790-86ad-dfd5f65f3a31
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 127 124 125
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:JuqU
a=ice-pwd:zlVuwktX7FdAJYbqRyP2QXCE
a=ice-options:trickle
a=fingerprint:sha-256 E8:82:F2:DC:B4:90:A9:EF:5D:F4:4B:8D:B5:51:CA:DB:EC:CD:93:A1:85:0A:20:80:89:35:43:72:DB:E9:19:25
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:- 1a22e567-5fa0-4048-8bbe-8d896140457a
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 AV1X/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:127 red/90000
a=rtpmap:124 rtx/90000
a=fmtp:124 apt=127
a=rtpmap:125 ulpfec/90000
a=ssrc-group:FID 627761831 1328663359
a=ssrc:627761831 cname:wWzqvgrPSbEVsUBN
a=ssrc:627761831 msid:- 1a22e567-5fa0-4048-8bbe-8d896140457a
a=ssrc:627761831 mslabel:-
a=ssrc:627761831 label:1a22e567-5fa0-4048-8bbe-8d896140457a
a=ssrc:1328663359 cname:wWzqvgrPSbEVsUBN
a=ssrc:1328663359 msid:- 1a22e567-5fa0-4048-8bbe-8d896140457a
a=ssrc:1328663359 mslabel:-
a=ssrc:1328663359 label:1a22e567-5fa0-4048-8bbe-8d896140457a
Why is Mediasoup not loading the h264 codec?
How could I get Mediasoup to work with WebRTC and h264 codec from iOS?
Thank you!
I had understood that there should only be the h264 codec.