Problem setting WebRTC connection between GStreamer and FreeSwitch

Tue Mar 27 13:48:39 UTC 2018

Hi,
I am trying to establish WebRTC connection between GStreamer and 
FreeSwitch. FreeSwitch itself works - I am able to connect to it using 
Blink VoIP client, and with WebRTC using Chrome+SIP.js.

I started with 
https://github.com/centricular/gstwebrtc-demos/blob/master/sendrecv/gst/webrtc-sendrecv.c 
example, and modified it to talk with JS script running on node.js, 
which uses SIP.js to communicate with FreeSwitch.

My pipeline is simplified to use audio only, and no STUN:
     pipe1 = gst_parse_launch("webrtcbin name=sendrecv "
         "audiotestsrc wave=red-noise ! queue ! opusenc ! rtpopuspay ! "
         "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ",
         &error);

This is SDP created by GST, with ICE candidates added:

v=0
o=- 8118924877098511175 0 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle
m=audio 9 UDP/TLS/RTP/SAVPF 97
c=IN IP4 0.0.0.0
a=setup:actpass
a=ice-ufrag:uAfKhCDTPsXL7o0pB1K2AZ6sTtqbmFt+
a=ice-pwd:RD0/zeks78Ix5jmeb7qKuPA14tRd1GNG
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:97 OPUS/48000/2
a=rtcp-fb:97 nack
a=rtcp-fb:97 nack pli
a=mid:audio0
a=fingerprint:sha-256 
CA:3D:48:EC:90:39:83:2C:C4:99:4B:1E:16:0A:6E:B4:82:A4:F9:8D:2A:E5:61:0C:E2:22:AA:3A:25:53:64:A3
a=candidate:10 1 UDP 2013266428 192.168.100.20 38669 typ host
a=candidate:11 1 TCP 1015022591 192.168.100.20 9 typ host tcptype active
a=candidate:12 1 TCP 1010828287 192.168.100.20 33547 typ host tcptype 
passive
a=candidate:10 2 UDP 2013266427 192.168.100.20 42780 typ host
a=candidate:11 2 TCP 1015022590 192.168.100.20 9 typ host tcptype active
a=candidate:12 2 TCP 1010828286 192.168.100.20 37994 typ host tcptype 
passive
a=end-of-candidates

FreeSwitch accepts it, and sends following one in reply:

v=0
o=FreeSWITCH 1522134060 1522134061 IN IP4 192.168.100.20
s=FreeSWITCH
c=IN IP4 192.168.100.20
t=0 0
a=msid-semantic: WMS 2TtM3ayYJHI44bvP2idZ8jCOcfQVNm4G
m=audio 19410 UDP/TLS/RTP/SAVPF 97 101
a=rtpmap:97 OPUS/48000/2
a=fmtp:97 useinbandfec=1; minptime=10; maxptime=40
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fingerprint:sha-256 
95:66:C4:94:4B:DD:8B:BD:06:29:DD:0B:96:9C:C2:D2:57:81:B3:61:8A:D9:4E:42:36:17:BB:9D:E3:BF:A8:B7
a=setup:active
a=rtcp-mux
a=rtcp:19410 IN IP4 192.168.100.20
a=ice-ufrag:lR5j1yY3DFPhRJNO
a=ice-pwd:WWDathYxuGwIKbOVTK5K9UDD
a=candidate:5323758004 1 udp 659136 192.168.100.20 19410 typ host 
generation 0
a=end-of-candidates
a=ssrc:1792117166 cname:6niZnGNXemS9odOg
a=ssrc:1792117166 msid:2TtM3ayYJHI44bvP2idZ8jCOcfQVNm4G a0
a=ssrc:1792117166 mslabel:2TtM3ayYJHI44bvP2idZ8jCOcfQVNm4G
a=ssrc:1792117166 label:2TtM3ayYJHI44bvP2idZ8jCOcfQVNm4Ga0

I send it in set-remote-description as in example code, but GST is not 
able to establish SRTP session. In Wireshark capture I see that 
FreeSwitch sent few STUN Binding Requests (GST responded to them) and 
RTCP packets with Receiver Report and Source description.

GST debug logs are very verbose and it was hard for me to find some 
useful info there. I tried to filter them a bit, and found following 
entries there:

0:00:01.611202908   542      0x1b5b4a0 WARN               structure 
gststructure.c:1832:priv_gst_structure_append_to_gstring: No value 
transform to serialize field 'srtp-key' of type 'buffer'

0:00:05.670806044   542      0x1bd1190 DEBUG         GST_SCHEDULING 
gstpad.c:4277:gst_pad_chain_data_unchecked:<dtlssrtpdec0:sink> calling 
chainfunction &gst_proxy_pad_chain_default with buffer buffer: 
0x7f6c10005040, pts 0:00:04.151691973, dts 0:00:04.151691973, dur 
99:99:99.999999999, size 92, offset none, offset_end none, flags 0x40
0:00:05.670825393   542      0x1bd1190 DEBUG         GST_SCHEDULING 
gstpad.c:4277:gst_pad_chain_data_unchecked:<dtlssrtpdemux0:sink> calling 
chainfunction &sink_chain with buffer buffer: 0x7f6c10005040, pts 
0:00:04.151691973, dts 0:00:04.151691973, dur 99:99:99.999999999, size 
92, offset none, offset_end none, flags 0x40
0:00:05.670856500   542      0x1bd1190 LOG            dtlssrtpdemux 
gstdtlssrtpdemux.c:129:sink_chain:<dtlssrtpdemux0> pushing rtp packet
0:00:05.671088870   542      0x1bd1190 WARN             dtlssrtpdec 
gstdtlssrtpdec.c:412:on_decoder_request_key:<dtlssrtpdec0> no srtp key 
available yet
0:00:05.671104066   542      0x1bd1190 WARN                 srtpdec 
gstsrtpdec.c:818:request_key_with_signal:<srtpdec0> Could not get caps 
for stream with SSRC 1792117166
0:00:05.671115242   542      0x1bd1190 WARN                 srtpdec 
gstsrtpdec.c:1262:gst_srtp_dec_chain:<srtpdec0> Invalid buffer, dropping
0:00:05.671154906   542      0x1bd1190 DEBUG         GST_SCHEDULING 
gstpad.c:4283:gst_pad_chain_data_unchecked:<dtlssrtpdemux0:sink> called 
chainfunction &sink_chain with buffer 0x7f6c10005040, returned ok
0:00:05.671166503   542      0x1bd1190 DEBUG         GST_SCHEDULING 
gstpad.c:4283:gst_pad_chain_data_unchecked:<dtlssrtpdec0:sink> called 
chainfunction &gst_proxy_pad_chain_default with buffer 0x7f6c10005040, 
returned ok

Any ideas how to fix this?

My FreeSwitch is installed on CentOS from official RPMs, and (mostly) 
with default config. GStreamer 1.14 is compiled from sources there.

Regards,
Daniel