SIPML5 and IPV6 Support

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DHAVAL INDRODIYA

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Nov 13, 2017, 11:12:01 AM11/13/17
to discuss-doubango
Dear All,

I am running demo for SIPML5 + Asterisk (14.5.0) without any issue and it is working for most of the personal computer.
Recently I noticed one issue with Apple Machine which network is in IPV6 and When I make a call from that machine We cannot here anything back from my asterisk server although signaling are fine and call also connected.

After call connected there is no voice from B-Leg of call on browser and no media packets are flowing from browser to asterisk server.

Here is First Invite Packet from SIPML5 on chrome browser with ICE candidate is filled with IPV6 Address.

<--- SIP read from WS:104.132.8.82:61084 --->
INVITE sip:11...@webrtc1.dhaval-webrtc.tk SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKeLcuHNBBG7vSSfuVfMKE6GTrqhqKk8D1;rport
From: "Spearline"<sip:60...@webrtc1.dhaval-webrtc.tk>;tag=uvkSTbqxZR46T27tsUAZ
To: <sip:11...@webrtc1.dhaval-webrtc.tk>
Contact: "Spearline"<sips:60...@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>;impi=6001;ha1=9597d55bc60f2ad74960de5a51f797a8;+g.oma.sip-im;language="en,fr"
Call-ID: 1bbfe9b4-f8dc-7fcc-3823-1336cce8953f
CSeq: 43091 INVITE
Content-Type: application/sdp
Content-Length: 1943
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=- 2748291604199996400 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS Lluu4EE8aT3qnD4zJENCKUMze8Dy9JfaOjUe
m=audio 53790 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 104.132.8.82
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2097826147 1 udp 2122262783 2620::1040:20:7cec:d059:c893:e390 53885 typ host generation 0 network-id 2 network-cost 10
a=candidate:3922505132 1 udp 2122194687 100.120.156.242 57673 typ host generation 0 network-id 1 network-cost 10
a=candidate:2147934829 1 udp 1685987071 104.132.8.82 53790 typ srflx raddr 100.120.156.242 rport 57673 generation 0 network-id 1 network-cost 10
a=candidate:864662931 1 tcp 1518283007 2620::1040:20:7cec:d059:c893:e390 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:2806993244 1 tcp 1518214911 100.120.156.242 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:eAPA
a=ice-pwd:t6yfje3Rm23a0IGQyBs51rbB
a=ice-options:trickle
a=fingerprint:sha-256 59:F1:77:56:79:77:EC:CD:1B:B0:59:87:5E:E1:F4:B2:8A:B0:35:05:22:B2:DC:43:77:48:6A:2F:80:39:B3:37
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
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



Now  If I dial from same browser same machine using IPv4 Network call connects and voice also passed.

Is it something we need to do with IPV6 Support ?

Do TURN server from numb.viaginie.ca solved this problem

Do I need to put my own TURN server to relay IPV6 and IPV4 packets ?

Will FreeSwitch solve this issue?

Do let me know if there is any other workaround.

Any help appreciated.
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