Comments (4)
Why don't you just put the STUN server's IPv4 address in the config, so that name resolution is not performed and you always get your v4 srflx candidates?
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Therefore I would like to make the proposal to resolve and connect to any STUN or TURN server via both v4 and v6 (if enabled) and add all responses to the address pool.
Unfortunately that's not possible, because we use libnice and you can only specify a single STUN server to use. The libnice library always expects an IP address for the STUN server to use, which is why we resolve any address ourselves if you provide a domain name, and we use the first result we get, which in your case probably is an IPv6 address. As such, the workaround Alex suggested would address your requirement.
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Thank you both for your responses.
I will probably go that way then, although it disables STUN usage for ipv6 completely (which is hopefully no problem in the field).
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although it disables STUN usage for ipv6 completely
You mentioned the public IPv6 IP you know already as it's local (which makes sense, it's what IPv6 was created for, not needing NATs), so I guess candidates for that address are already gathered. Closing the issue.
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Related Issues (20)
- [1.x] Sip session get stuck HOT 12
- No call back triggering in 'onlocalstream' HOT 2
- [1.x] Streaming plugin does not send "started" event to event handlers HOT 2
- Add unauthenticated no-payload ping endpoint for health checks HOT 3
- Build fails with latest libsrtp version HOT 9
- [0.x] [janus.js] black screen or streaming stops periodically when connected through WebRTC (Chromium only) HOT 4
- [1.x] Issue with outbound calls for SIP Gateway HOT 3
- [1.x] videoroom: remote publisher doesn't release RTP/RTCP ports HOT 10
- [1.x] Does AudioBridge supports multiple participant join from RTP in a single room . Getting `Already in a room` Error for 2nd participant HOT 3
- [1.x] rtp_port_range not being respected HOT 2
- [1.x] Crash / segfault on 1.2.2 HOT 5
- [1.x] system under heavy load stops to handle calls wia ws api HOT 16
- [1.x] Crash / segfault on 1.2.3 HOT 10
- [1.x] inconsistency happen when more than 6 users join at same time HOT 2
- [1.x]inconsistency happen when more than 6 users join at same time with different browsers (web and mobile) HOT 8
- [1.x] videoroom: support of string_ids/string_ids_user HOT 2
- [1.x] Increased event handlers SIP logging buffer size one more time or make it configurable HOT 5
- [1.x] build failure on macos HOT 3
- [0.x] Possible memory leaks found with libasan latest HOT 12
- [1.x] SIP plugin "Couldn't connect audio RTP? -- 22 (Invalid argument)" HOT 5
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