Top 5 Challenges To Add Web Calls to Truphone VoIP Platform

Top 5 Challenges to Add
Web Calls to Truphone VoIP
Platform
Giacomo Vacca
Senior Network Applications Developer

© 2014 Truphone Limited. All Rights Reserved.05 August 2014
labs.truphone.com
2

© 2014 Truphone Limited. All Rights Reserved.05 August 2014 3
WebRTC is not for VoIP devs

The “Traditional Architecture”

• Follow standards
• Use Open Source products
• Think scalability since the beginning
• Don’t compromise on security
• Change often, keep technical debt low
• Deploy automatically
5
Design Principles

• JS client (JsSIP-based)
• WebSockets
• Authentication
• Media relay for p2p sessions
• FreeSWITCH out of the media path
• Web/VoIP/PSTN interaction
–Support for DTLS
6
Changes Required

Websocket support
7
Challenge 1

Adding WebSocket support…

… with Kamailio

Kamailio websocket module
loadmodule "websocket.so"
…
event_route[xhttp:request] {
set_reply_close();
set_reply_no_connect();
…
if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade"
&& $rm=~"GET") {
if (ws_handle_handshake()) {
exit;
}
}
}
10
WebSockets support

v=0
o=- 2700277954018656518 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS e8hFvtt1SQ9FiqpbYbOhN19wvpp21flszxZ4
m=audio 63529 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 162.243.255.49
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
11
Gigantic SDP 1/10

a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=rtcp:63529 IN IP4 162.243.255.49
a=candidate:1867667642 1 udp 2122260223 192.168.254.1 58603 typ
host generation 0
host generation 0
host generation 0
host generation 0
12
Gigantic SDP 2/10

a=candidate:882550992 1 udp 2122129151 10.2.129.202 58605 typ host
generation 0
generation 0
a=candidate:567387210 1 tcp 1518280447 192.168.254.1 0 typ host
generation 0
generation 0
generation 0
generation 0
generation 0
13
Gigantic SDP 3/10

generation 0
a=candidate:3427568147 1 udp 41754367 162.243.255.49 63529 typ relay
raddr 195.110.37.225 rport 48569 generation 0
a=ice-ufrag:JuWsZDrjSjaP91bz
a=ice-pwd:QUc3SJ/aJe+FLa658U5y2qJp
a=ice-options:google-ice
a=fingerprint:sha-256
C0:42:EE:16:BF:BD:87:76:A4:0F:F6:A9:48:C8:BA:45:2E:0E:AD:1F:41:AA:
C3:3C:99:45:CA:FE:50:9D:E4:32
a=setup:actpass
a=mid:audio
14
Gigantic SDP 4/10

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=rtcp-mux
a=maxptime:60
a=ssrc:2566549015 cname:TdImtoO7ee/QmnJN
a=ssrc:2566549015 msid:e8hFvtt1SQ9FiqpbYbOhN19wvpp21flszxZ4
d7516c7a-7f81-4598-a2e6-aa71b7ceb8db
a=ssrc:2566549015 mslabel:e8hFvtt1SQ9FiqpbYbOhN19wvpp21flszxZ4
a=ssrc:2566549015 label:d7516c7a-7f81-4598-a2e6-aa71b7ceb8db
m=video 63529 RTP/SAVPF 100 116 117 96
c=IN IP4 162.243.255.49
a=rtpmap:100 VP8/90000
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
15
Gigantic SDP 5/10

a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=rtcp:63529 IN IP4 162.243.255.49
host generation 0
host generation 0
host generation 0
host generation 0
generation 0
generation 0
16
Gigantic SDP 6/10

generation 0
generation 0
generation 0
generation 0
generation 0
generation 0
17
Gigantic SDP 7/10

a=ice-ufrag:JuWsZDrjSjaP91bz
a=ice-pwd:QUc3SJ/aJe+FLa658U5y2qJp
a=ice-options:google-ice
a=fingerprint:sha-256
C0:42:EE:16:BF:BD:87:76:A4:0F:F6:A9:48:C8:BA:45:2E:0E:AD:1F:41:AA:
C3:3C:99:45:CA:FE:50:9D:E4:32
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=rtcp-mux
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
18
Gigantic SDP 8/10

a=ssrc-group:FID 1964375756 3405612694
fb4cdb67-4167-4f63-a934-fa49977f955a
a=ssrc:1964375756 label:fb4cdb67-4167-4f63-a934-fa49977f955a
fb4cdb67-4167-4f63-a934-fa49977f955a
a=ssrc:3405612694 label:fb4cdb67-4167-4f63-a934-fa49977f955a
19
Gigantic SDP 9/10

Gigantic SDP 10/10
Increase tcp_rd_buf_size!

WebRTC/VoIP and WebRTC/PSTN
21
Challenge 2

• FreeSWITCH to the rescue
– session:execute("export", "media_webrtc=true")
– session:setVariable("bypass_media", "true")
• Certs for DTLS
– /etc/freeswitch/tls/dtls-srtp.crt
• Transcoding (mod_opus)
– <load module="mod_opus"/>
22
WebRTC/VoIP (and PSTN)

WebRTC/VoIP - WebRTC/PSTN

Authentication
24
Challenge 3

Ephemeral Authentication

Kamailio auth_ephemeral module
loadmodule "auth.so“
loadmodule "auth_ephemeral.so“
…
modparam("auth_ephemeral", "secret", AUTH_EPHEMERAL_SECRET)
…
if (!autheph_proxy("$fd")) {
auth_challenge("$fd", "0");
exit;
}
26
Ephemeral Authentication

Media relay for p2p
27
Challenge 4

• rfc-5766-turn-server
–Open Source
–Easy to set up (e.g. Puppet)
–Ephemeral auth. applies here too!
• JsSIP configuration easy:
– turn_servers: { urls: [“turn:myturn.com”],
username: “u”, credentials: “c”}
28
Media relay (TURN + STUN)

Media relay (TURN + STUN)

Testing
30
Challenge 5

• From sipp & PJSUA to… Selenium?
• Chrome and FF change
–A lot
–Often
• JsSIP master vs develop (re-INVITEs)
• FreeSWITCH 1.5 or 1.4?
31
5. Testing

Dealing with Web Developers 
32
Bonus challenge

• Different tools/processes
• Bleeding edge libs/apps
• Automating deployment of non-stock apps?
• Don’t care about signalling/media
33
Web Development

findable.io

• Big shift to web technologies
• No need to revolutionize your VoIP platform
• SIP can do the job, but…
–verto endpoint for FreeSWITCH?
• (Automated) Testing is hard
• chrome://webrtc-internals is cool
35
Wrapping up

Q&A
Giacomo Vacca - @giavac
labs@truphone.com
https://labs.truphone.com/about/
36

Top 5 Challenges To Add Web Calls to Truphone VoIP Platform

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