Problem setting WebRTC connection between GStreamer and FreeSwitch

Wed Mar 28 08:27:26 UTC 2018

On 28/03/18 00:48, daniel at poradnik-webmastera.com wrote:
> 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

This a=rtcp line is problematic if FreeSwitch is expecting RTCP packets
to appear on port 19410.  RTCP is transmitted through the ICE connection
and thus is not generally signalled through the SDP.

> 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

Does FreeSwitch require the msid/cname handling because that's currently
not implemented by webrtcbin.

> 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

This usually means that something's not quite right with what GStreamer
expects.

> Any ideas how to fix this?

Depends very much on what FreeSwitch is expecting and producing :).

Cheers
-Matt

> My FreeSwitch is installed on CentOS from official RPMs, and (mostly)
> with default config. GStreamer 1.14 is compiled from sources there.
>
> Regards,
> Daniel
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

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