I have a working telephone project that uses SIP.js 0.7.5 with Asterisk on the
server side. Currently it handles both audio and video correctly.
The SIP.js webpage has instructions for setting up a datachannel through a SIP
call. The online demo uses OpenSIPS.
When setting up a SIP call with a datachannel through the SIP.js online demo,
the SDP looks like this:
v=0
o=- 2849011408178506361 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS
m=application 45029 DTLS/SCTP 5000
c=IN IP4 201.234.196.170
a=candidate:1184643355 1 udp 2122260223 10.1.0.1 33877 typ host generation 0
network-id 3
a=candidate:2680228395 1 udp 2122194687 10.0.0.245 49916 typ host generation 0
network-id 2
a=candidate:3700057831 1 udp 2122129151 192.168.3.2 45029 typ host generation 0
network-id 1
a=candidate:136300011 1 tcp 1518280447 10.1.0.1 9 typ host tcptype active
generation 0 network-id 3
a=candidate:3510826715 1 tcp 1518214911 10.0.0.245 9 typ host tcptype active
generation 0 network-id 2
a=candidate:2450102807 1 tcp 1518149375 192.168.3.2 9 typ host tcptype active
generation 0 network-id 1
a=candidate:1976953138 1 udp 1685921535 201.234.196.170 45029 typ srflx raddr
192.168.3.2 rport 45029 generation 0 network-id 1
a=ice-ufrag:8Dt8
a=ice-pwd:CPt/905Ehc9EKsY+Dpu4UQ7d
a=fingerprint:sha-256
20:46:82:E4:29:DE:EB:9A:04:74:A4:31:F7:A5:7A:45:15:8B:9C:61:50:CB:8A:EE:2E:35:5A:74:10:33:09:B3
a=setup:actpass
a=mid:data
a=sctpmap:5000 webrtc-datachannel 1024
a=sendrecv
a=candidate:0dspPHO0b1Ifs6Nv 1 UDP 16777215 199.7.173.73 56338 typ relay raddr
199.7.173.73 rport 56338
The media m= is of type application.
Is Asterisk capable of handling such a SDP, so that two SIP endpoints registered through Asterisk can begin exchanging data? From what I understand in the code, Asterisk will reject such a call. However, I want to exhaust what Asterisk can do before
resorting to setting up a SIP proxy between the endpoint and Asterisk. I remember reading a report about a webrtc success story where the webphones were also exchanging data using datachannels.
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