#35 WebRTC calls are dying due to authentication failure

[David Suendermann-Oeft]

Some calls are dying in the middle during some of the longer interactions.

Vikram and I placed a number of calls, and this is what is happening: About 10 to 20 seconds into every call, it dies coinciding with the following type of error messages I am seeing in Asterisk:

[Apr 30 19:20:09] NOTICE[20547]: chan_sip.c:27851 handle_request_register: Registration from '"7005" sip:7005@54.173.240.119:5060' failed for '107.150.39.86:5083' - Wrong password
-- Executing [i@default:1] background("SIP/JVXMLcloud1-0000000a", "demo-instruct")
-- <sip jvxmlcloud1-0000000a=""> Playing 'demo-instruct.ulaw' (language 'en')</sip>

[Apr 30 19:34:20] NOTICE[21485]: chan_sip.c:27851 handle_request_register: Registration from '"489" sip:489@54.173.240.119:5060' failed for '212.83.153.159:5076' - Wrong password

[Apr 30 19:48:27] NOTICE[21588]: chan_sip.c:27851 handle_request_register: Registration from '"100" sip:100@54.173.240.119:5060' failed for '212.83.153.159:5067' - Wrong password
[Apr 30 19:48:43] WARNING[21613][C-00000000]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10

These are apparently crawlers which try to get in our system from the outside world. I do not know why a failed authentication attempt makes the Halef connection drop; I did not experience such issue in Stuttgart.

[David Suendermann-Oeft]

The problem persists:

[May 1 17:22:38] WARNING[27581][C-00000003]: chan_iax2.c:1220 jb_warning_output: Resyncing the jb. last_delay 1, this delay 9408, threshold 1002, new offset -21496
[May 1 17:22:53] WARNING[27579][C-00000003]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10

[May 1 17:46:18] WARNING[27766][C-00000005]: chan_iax2.c:1220 jb_warning_output: Resyncing the jb. last_delay -1, this delay 13625, threshold 1002, new offset -35226
[May 1 17:46:20] WARNING[27764][C-00000005]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10

[May 1 17:50:21] WARNING[27774][C-00000006]: chan_iax2.c:1220 jb_warning_output: Resyncing the jb. last_delay 0, this delay 8459, threshold 1002, new offset -30263
[May 1 17:50:26] WARNING[27772][C-00000006]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10

There is quite a number of forum chat about this error, one of which also covers the Firefox v37 situation:

http://community.freepbx.org/t/webrtc-with-asterisk-13-1-and-freepbx-12/26486/5

One of the forum discussions suggests it is a browser-specific issue:

http://forums.digium.com/viewtopic.php?f=13&t=87590

So, I tried it with Chrome. Surprisingly, Chrome is currently working from the office. And, yes, it did not die after receiving the authentication failure:

[May 1 18:01:02] WARNING[27782][C-00000007]: chan_iax2.c:1220 jb_warning_output: Resyncing the jb. last_delay -1, this delay 8845, threshold 1002, new offset -23015
[May 1 18:01:12] WARNING[27782][C-00000007]: chan_iax2.c:1220 jb_warning_output: Resyncing the jb. last_delay 1, this delay 10656, threshold 1002, new offset -33671
-- Got SIP response 500 "JsSIP Internal Error" back from 144.81.164.9:5060
> 0x7fd80000bdf0 -- Probation passed - setting RTP source address to 144.81.164.9:24079
[May 1 18:01:16] WARNING[27780][C-00000007]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
[May 1 18:01:26] WARNING[27782][C-00000007]: chan_iax2.c:1220 jb_warning_output: Resyncing the jb. last_delay -1, this delay 13638, threshold 1002, new offset -47309
[May 1 18:01:54] WARNING[27782][C-00000007]: chan_iax2.c:1220 jb_warning_output: Resyncing the jb. last_delay 0, this delay 27551, threshold 1002, new offset -74860

But then, I am only able to hear the first system prompt! Chrome does not currently play back the other prompts. Oh my...

Yours,

DSO

[David Suendermann-Oeft]

We found that this issue causes the call to die only in JSSIP, but not in SIPML5.