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| 00:29.21 | *** mode/#asterisk-dev [+o mjordan] by ChanServ |
| 00:29.41 | file | ahhhhhhhhhhhhh! mjordan! |
| 00:29.48 | mjordan | run! |
| 00:30.14 | file | I decline. |
| 00:30.33 | mjordan | that's probably wise. |
| 00:30.37 | mjordan | tis cold out |
| 00:30.59 | file | I was talking to my brother and apparently... it is quite a bit warmer in Newfoundland |
| 00:31.05 | mjordan | water |
| 00:31.14 | file | I don't care. It should be colder! |
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| 03:17.21 | tm1000 | file: are there any known bugs with webrtc in asterisk11? |
| 03:17.27 | tm1000 | im having....issues... |
| 03:17.45 | mjordan | tm1000: yes, but that's one of those "it depends" kind of answers |
| 03:17.46 | tm1000 | internal calls and echo tests are fine. |
| 03:17.58 | tm1000 | mjordan: yeah its really weird, like really weird |
| 03:18.04 | mjordan | which browser? |
| 03:18.19 | tm1000 | chrome, the only decent webrtc one (that doesnt require certs) |
| 03:18.26 | mjordan | k, firefox I know has more issues |
| 03:18.28 | tm1000 | if I dial out over my trunk its one way audio land, |
| 03:18.34 | tm1000 | but here's the weird thing |
| 03:18.50 | tm1000 | if I dail out, I only hear the recieving party |
| 03:18.58 | tm1000 | If I dail in, I only hear my laptop |
| 03:19.12 | tm1000 | so it switches the one way signal as well |
| 03:19.22 | mjordan | is this 12? |
| 03:19.23 | tm1000 | but like I said, endpoint <--> endpoint is fine |
| 03:19.24 | tm1000 | nop |
| 03:19.26 | mjordan | k |
| 03:19.28 | tm1000 | no this is 11.7 |
| 03:19.30 | mjordan | (you said that, sorry) |
| 03:19.48 | tm1000 | last time I checked this was 11.6/5 in october |
| 03:20.17 | mjordan | hm. That is weird. What are the directmedia settings? And for endpoint to endpoint, are both over WS? |
| 03:20.39 | tm1000 | If I was really having a one-way audio issue I would expect it to be consistent |
| 03:20.57 | tm1000 | no they are not both WS, but I thought the WS <--> WS was only a requirement in 12 |
| 03:21.32 | tm1000 | let me check one more time with UDP <--> WS to make sure im not on crazy crack |
| 03:21.37 | mjordan | no, just trying to figure out the bridging |
| 03:21.42 | tm1000 | as *43 works (*43 is echo test in fpbx) |
| 03:22.08 | tm1000 | but the echo test would be WS <--> internal asterisk so let me check with another ext again |
| 03:23.44 | tm1000 | mjordan: ok yeah, Bria (UDP) <--> WS works (internal) |
| 03:24.04 | tm1000 | its just when I route to an outbound state |
| 03:24.11 | tm1000 | or if I get an inbound call and answer it |
| 03:24.26 | mjordan | So, ASTERISK-23145 is the closest I know of |
| 03:24.47 | mjordan | see, the strange thing is that your outbound is SIPStation, right? |
| 03:24.47 | tm1000 | it happens on two separate machines as well on completely different networks |
| 03:24.49 | tm1000 | looks |
| 03:24.54 | tm1000 | mjordan: yes to both |
| 03:24.59 | tm1000 | err for both systems |
| 03:25.18 | tm1000 | though… I can force it over an old wildcard TDM |
| 03:25.22 | tm1000 | if you want me to try |
| 03:25.25 | mjordan | nah |
| 03:25.29 | mjordan | at that point, the RTP stops |
| 03:25.42 | mjordan | So it's a pretty different situation |
| 03:26.02 | tm1000 | also of note, I have both sipml5 and jssip |
| 03:26.10 | tm1000 | and I can switch between the two and the issue is the same |
| 03:26.32 | mjordan | in your situation, going WS to UDP works (extension to extension)? |
| 03:26.54 | tm1000 | yes |
| 03:26.59 | tm1000 | every time |
| 03:27.21 | tm1000 | its just when, well I want to say bridging is involved, but thats incorrect because its bridging the exts as well |
| 03:27.42 | tm1000 | its just if I route over a trunk |
| 03:29.35 | mjordan | yeah. I almost want to say it's probably a local_net or NAT setting messing it up |
| 03:29.54 | mjordan | one of the few things that did change in the last few versions of 11 were some NAT settings, specifically with rport and some other settings getting applied |
| 03:30.24 | mjordan | I wonder if they're mucking up the WS connection somehow - because that's probably the only difference between your extensions and your trunk |
| 03:31.01 | mjordan | tm1000: mind making an issue? Throw a debug log with sip set debug on it as well if you don't mind |
| 03:31.22 | mjordan | This sounds similar-ish to Rusty's issue |
| 03:31.28 | tm1000 | mjordan: yeah. I wanted to check with you guys first |
| 03:31.31 | mjordan | but since that was 12 and pjsip, it's pretty different |
| 03:31.32 | tm1000 | and do more testing |
| 03:31.36 | mjordan | yeah, I think it's something that crept in |
| 03:31.44 | tm1000 | as Im trying to get it to work through the sipml5 interface |
| 03:32.09 | mjordan | we've been ironing out the NAT stuff recently, and it's possible that we fixed something with everything normal that is mucking up that transport |
| 03:32.17 | tm1000 | which if it works in the sipml5 demo page then my bad |
| 03:32.26 | mjordan | shrugs |
| 03:32.29 | mjordan | it's worth us taking a look |
| 03:33.18 | tm1000 | well shoot |
| 03:33.23 | tm1000 | it works through the demo page |
| 03:35.32 | tm1000 | mjordan: ok now it works on my page |
| 03:35.39 | tm1000 | this shit is rediculous |
| 03:35.45 | tm1000 | throws table at webrtc |
| 03:35.50 | tm1000 | I feel bad for you and file |
| 03:36.03 | tm1000 | I didnt change one thing... |
| 03:37.22 | tm1000 | mjordan: ok its definately the issue you sent me |
| 03:42.03 | tm1000 | now I cant replicate it |
| 03:47.44 | mjordan | lawl |
| 03:48.12 | mjordan | we'll have to investigate what sipml5 is sending us. It may be a bug there |
| 03:48.14 | mjordan | shrugs |
| 03:48.19 | mjordan | \o/ for moving targets |
| 03:48.38 | mjordan | ponders |
| 03:48.48 | coreyf1513 | i'm glad nobody has asked me to support webrtc yet, dreading the day |
| 03:48.58 | mjordan | the weekend project may be resurrecting our bots. I need some ascii art for flipping a table |
| 03:49.09 | mjordan | coreyf1513: if you control the environment, I think you could do some amazing things |
| 03:49.21 | mjordan | call centers should be *running* to that |
| 03:49.33 | mjordan | things get wonky quickly on ye olde interwebs |
| 03:49.40 | coreyf1513 | running to webrtc? |
| 03:49.54 | mjordan | yes, if you control the environment - that is, the browser, the stack, etc. |
| 03:50.14 | mjordan | no deploying soft phones, integrate it into your call center as you see fit, etc. |
| 03:50.20 | mjordan | There's a lot of benefits for that situation |
| 03:50.44 | mjordan | Browser/stack versions have been problematic |
| 03:51.04 | coreyf1513 | I distribute a software based agent software based on xulrunner (mozilla), it includes a modified copy of pjsua that handles the actual SIP/media stuff |
| 03:51.43 | coreyf1513 | not as good as having the softphone in-process with the UI, but it was originally created a few years ago before webrtc had any implementations |
| 03:51.52 | mjordan | but the vast majority of issues have been in the browsers trying to handle NAT and mobile environments, and people trying to use WebRTC over the internet have had mixed results. Sometimes awesome (same browser -> same browser), sometimes not |
| 03:52.11 | tm1000 | mjordan: same issue as rusty now. its hit and miss with both sipml5 and pjsip |
| 03:52.12 | mjordan | coreyf1513: beats having to install a softphone all over the place |
| 03:52.17 | tm1000 | do you want a pcap as well |
| 03:52.19 | mjordan | tm1000: crazy. |
| 03:52.26 | mjordan | tm1000: nah, just chime in on rusty's issue |
| 03:52.28 | tm1000 | i did the sip debug about to upload it to a new ticket... |
| 03:52.29 | mjordan | I think he had a log file |
| 03:52.33 | mjordan | that'd be great |
| 03:52.33 | tm1000 | ah ok ill go on rustys |
| 03:52.36 | mjordan | we'll link em together |
| 03:52.45 | coreyf1513 | ahhh, so the fact that my call center agents are always on local IP's or VPN means I wouldn't have issues with webrtc? |
| 03:52.47 | tm1000 | mjordan: err together or separate :-p |
| 03:53.18 | mjordan | coreyf1513: you might still have issues, but most of the one's I've seen have been ICE/interop due to various versions of things |
| 03:53.26 | mjordan | tm1000: either! |
| 03:53.28 | mjordan | I'm flexible |
| 03:53.37 | tm1000 | coreyf1513: its purely gimicky for us… oh look webrtc cool…but dont actually use it for anything productive....oh |
| 03:53.46 | mjordan | it isn't ready yet. |
| 03:54.30 | tm1000 | mjordan: file told me the mailing list for webrtc is pretty depressing |
| 03:54.36 | mjordan | very. |
| 03:54.49 | mjordan | I follow some of the IETF discussions. |
| 03:54.55 | mjordan | I think Tim's post on SDP was about dead on |
| 03:55.16 | mjordan | http://webrtchacks.com/tim-rant/ |
| 03:55.33 | tm1000 | reads |
| 03:55.41 | mjordan | that SDP is glorious |
| 03:55.51 | mjordan | a=msid-semantic: WMS |
| 03:55.55 | mjordan | W.T.F. |
| 03:59.21 | tm1000 | mjordan: btw, even though there is one way audio, RTP debug shows streams going both ways |
| 03:59.26 | tm1000 | and asterisk never disconnects |
| 03:59.28 | mjordan | gah |
| 03:59.31 | tm1000 | even after 3min |
| 04:00.02 | mjordan | I'm leaning more towards sipml5 doing *something* then, particularly if the RTP looks to be going to the right addresses |
| 04:00.20 | mjordan | We might have to break out JsSIP |
| 04:00.49 | mjordan | has faith in Inaki |
| 04:01.14 | mjordan | tm1000: you know Qwell wrote the websocket transport for pjsip |
| 04:01.19 | mjordan | is just sayin |
| 04:08.57 | tm1000 | mjordan: im using jsSip |
| 04:09.00 | tm1000 | not sipml5 |
| 04:09.17 | tm1000 | I have the code branched in git, so I can switch between the two in a second |
| 04:09.19 | tm1000 | :-) |
| 04:15.22 | tm1000 | attached |
| 04:16.03 | tonyclewis | ya to webrtc |
| 04:16.14 | tonyclewis | they best thing to ever happen for communications |
| 04:16.16 | tonyclewis | LOL |
| 04:16.31 | tonyclewis | mjordan: we are pushing out the FreePBX WebRTC module this weekend |
| 04:16.35 | tonyclewis | ahead of ITExpo |
| 04:16.40 | tonyclewis | so expect lots of bugs and reports |
| 04:16.41 | tonyclewis | LOL |
| 04:16.58 | tonyclewis | mjordan: BTW over 500 installs of Asterisk 12 with FreePBX 12 |
| 04:17.02 | tonyclewis | and still no bugs |
| 04:17.05 | tonyclewis | that is really scary |
| 04:17.38 | tm1000 | tonyclewis: I think people are just doing "oh thats cool" and then go back |
| 04:17.53 | mjordan | I might agree with tm1000 |
| 04:17.58 | mjordan | I mean, I'm finding bugs |
| 04:18.40 | mjordan | people "know" us probably too well at this point. They'll all wait for 12.2 or 12.3 |
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| 04:19.26 | tonyclewis | hehe mjordan |
| 04:19.33 | tonyclewis | or 12.6 |
| 04:19.46 | tonyclewis | but I would of expected some reports by now of things |
| 04:19.54 | tonyclewis | as we know things are still broke |
| 04:20.04 | mjordan | yeah :-\ |
| 04:20.16 | mjordan | we'll cut 12.1-rc1 next week |
| 04:20.30 | mjordan | should have a number of things rolled into it by that point, lots of code got reviewed this week |
| 04:20.32 | tonyclewis | tm1000: I was able to get a load on our jenkins server just now upto 460 |
| 04:20.34 | tonyclewis | hehehe |
| 04:20.44 | tonyclewis | mjordan: cool |
| 04:20.48 | tonyclewis | as our 12.0 RPMs |
| 04:20.59 | tonyclewis | are just a bastard of patches and hacks and pulls from SVN |
| 04:21.03 | tm1000 | tonyclewis: what is wrong with jenkins and web2 |
| 04:21.05 | tonyclewis | be nice to get that cleaned up |
| 04:21.16 | tonyclewis | I had like 25 mockroots running |
| 04:21.18 | tm1000 | they've both been doing the high load thing recently.. |
| 04:21.19 | tonyclewis | building things |
| 04:21.21 | tm1000 | ah |
| 04:22.06 | tonyclewis | going to build a new Distro composing server all on SCSI drives |
| 04:22.10 | tonyclewis | help bring down I/Os |
| 04:22.32 | tonyclewis | its all binding on the I/Os |
| 04:27.03 | tm1000 | tonyclewis: web3 is doing great after you moved that around to its own machine |
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| 22:34.29 | mjordan | ASTERISK-1234 |
| 22:34.29 | CaptainPants | [ASTERISK-1234] [Status: Closed] IAX taking down linked boxes - https://issues.asterisk.org/jira/browse/ASTERISK-1234 |
| 22:34.33 | mjordan | cheers |
| 22:34.50 | mjordan | CaptainPants: hello |
| 22:36.15 | WIMPy | Oh, nice. I think I've seen that happening. |
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| 23:06.56 | mjordan | sighs |
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| 23:34.56 | mjordan | a little jumpy, eh CaptainPants |
| 23:34.59 | mjordan | yes, yes he is |
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| 23:36.18 | Fwny | Does anyone know if there is a reason ssrc changes when involving SRTP on a bridged call? |
| 23:40.27 | Fwny | Trying to get Asterisk 12 to play nice with WebRTC. Tracked it down to ssrc changing sometime after the INVITE comes in (ssrc needs to stay the same or it breaks Chrome's media implementation) |
| 23:41.34 | Fwny | patched it out in res_rtp_asterisk.cc with https://issues.asterisk.org/jira/secure/attachment/44054/p2pbridge.diff |
| 23:41.44 | Fwny | and it works when I do the call without crypto |