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| 02:11.52 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.32.0 (2020/03/12) 16.9.0 (2020/03/12) Standard: 17.3.0 (2020/03/12); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 15:35.53 | MLC | I've been pondering the following idea. a) Is it feasible? b) Is it a good idea? Idea: Leave one asterisk server in-house as a termination point for our POTS land lines and move the main asterisk server to the cloud. Maybe eventually port the POTS lines to VOIP. Goals: 1) Eliminate MOH bandwidth usage to our office for calls waiting in the queue. 2) Take advantage of reduced latency between asterisk server and voip.ms. 3) Take advant |
| 15:35.53 | MLC | age of reduced latency and increased stability between asterisk and our application server. (We have a number of integrations between the two.) |
| 15:41.51 | Samot | How does the MoH usage play into this? |
| 15:41.56 | Samot | Are you streaming it? |
| 15:42.27 | MLC | Not streaming, but it uses up RTP bandwidth between our office and voip.ms and a caller is waiting in the queue. |
| 15:42.50 | MLC | like right now, when we have 30+ waiting |
| 15:44.34 | Samot | The amount of RTP bandwidth being used for 30 calls is the same regardless of where they are. |
| 15:44.42 | Samot | Queue or on a call. |
| 15:44.49 | Samot | It's the same about of bandwidth. |
| 15:45.04 | MLC | yes, but if the phone server is in the cloud, it doesn't use up the bandwidth to our office, it is only used in the cloud. |
| 15:45.18 | MLC | Only need to use the bandwidth to the office when the call connect to an agent |
| 15:45.23 | Samot | Sure. |
| 15:45.28 | Samot | I'm not saying that's not a bad idea. |
| 15:45.32 | Samot | You just wrapped in the MoH |
| 15:45.41 | Samot | And RTP bandwidth due to it. |
| 15:45.43 | MLC | understood |
| 15:49.56 | MLC | What would I do to pass the inbound calls on the POTS lines from the in-house server to the asterisk in the cloud? |
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| 15:54.04 | igcewieling | That should happen by default in most cases. |
| 15:55.55 | MLC | Right now those POTS lines enter at a context in my dial plan that puts them in the queue. I assume that I need to do something to send those calls from that asterisk server to the one in the cloud? |
| 15:56.27 | igcewieling | Ah, now a queue is involved. I didn't realize that. |
| 15:56.50 | MLC | Just to keep things interesting! |
| 15:57.06 | igcewieling | Without a queue, it would be something like Dial(PJSIP/peername/12125551212) |
| 15:57.33 | igcewieling | Anyway, I thought the queue was on the cloud server. |
| 15:59.19 | MLC | And I would set up the cloud server and endpoint in the pjsip.conf in the in-house server? |
| 15:59.26 | MLC | *as an endpoint |
| 15:59.38 | igcewieling | of course. the remote server is an endpoint |
| 15:59.49 | MLC | makes sense |
| 16:00.25 | Samot | OK |
| 16:00.33 | igcewieling | Your calls will be something like POTS -> localasterisk server -> hosted asterisk server -> ???? -> phone |
| 16:00.47 | Samot | Why would you move the system to the cloud to save on bandwidth but then route calls back through the on-prem to the cloud back to the agents? |
| 16:01.52 | MLC | Samot: a) saves the MoH bandwidth when callers are in the queue. b) less latency to voip.ms server and our application server. |
| 16:02.13 | igcewieling | MLC, not on your pots lines |
| 16:02.14 | MLC | Most of our inbound calls are VOIP, not POTs |
| 16:02.24 | MLC | POTS is maybe 10% of the traffic |
| 16:02.57 | MLC | Would probably port the POTS to VOIP in this scenario |
| 16:03.08 | igcewieling | can't you just call forward the pots lines to voip doing something like *72? |
| 16:04.57 | MLC | Maybe. Good idea. I had not thought about that. |
| 16:05.40 | igcewieling | Trying to keep pots lines when moving to "the cloud" is a dumb idea. |
| 16:05.51 | MLC | Agreed |
| 16:07.10 | Kobaz | ugh, linphone really doesn't do well with tcp/tls |
| 16:07.41 | igcewieling | All softphones suck. |
| 16:07.50 | Kobaz | i know |
| 16:07.56 | Kobaz | microsip seems to handle it better |
| 16:08.06 | Samot | MLC: My point is, call comes in over POTS to on-prem, routes to cloud for the queue, you now have that going over the bandwidth |
| 16:08.25 | Samot | 12:07:41 PM <igcewieling> All softphones suck. <-- This is highly incorrect. |
| 16:08.26 | igcewieling | hugs his Polycom VVX 6xx with a sidecar w 32 line appearances. |
| 16:08.44 | Kobaz | Samot: there are soft phones that don't suck? |
| 16:08.45 | Samot | Again, and I'll keep saying this, I use the Bria and it works. |
| 16:09.18 | igcewieling | Fine. All free softphones suck, most non-free softphones suck. |
| 16:10.29 | igcewieling | Kobaz: if you consider all the time you've spent on this, you could have purchased an inexpensive linksys/sipura/whaevertheyarecalledtoday instead and still come out ahead. |
| 16:10.39 | Kobaz | no |
| 16:10.40 | Kobaz | haha |
| 16:10.47 | MLC | Samot: I agree totally. Moving from a voice T1 to VOIP was a big step for our management, so I'm taking it a bit at a time. |
| 16:10.52 | Kobaz | this is for like hundreds of people working from home |
| 16:10.58 | Kobaz | i had one customer buy zoiper |
| 16:11.00 | MLC | +1 for Bria |
| 16:11.04 | Kobaz | pretty decent results so far |
| 16:11.23 | Samot | I think I said this the other day during one of the all softphone sucks rants. |
| 16:11.37 | Samot | Having Bria Enterprise was a blessing for what has happened. |
| 16:11.54 | Samot | Push services, provisioning, multi device setups for users. |
| 16:12.05 | Kobaz | that's about like 800 a month, rihgt? |
| 16:12.09 | Kobaz | something like that |
| 16:12.09 | Samot | Nope. |
| 16:12.23 | Samot | Per license. |
| 16:12.26 | Kobaz | ah |
| 16:12.48 | igcewieling | $2.95/month according to a a quick google search. |
| 16:12.52 | Kobaz | ah |
| 16:12.59 | Samot | Depends on the license. |
| 16:14.30 | Samot | Also during all of this Bria Enterprise has 90 day licenses for now. |
| 16:14.56 | Samot | So if I needed a temp license for the user that would never normally use Bria outside of being force to WFH, I can get 90 days. |
| 16:16.43 | igcewieling | for 100 users, that is only $3,588 /year |
| 16:20.04 | igcewieling | Kobaz: the "Zulu" softphone included with the distro isn't good enough for you? |
| 16:20.32 | Kobaz | i duno, never used it |
| 16:20.41 | Kobaz | included in what distro |
| 16:20.53 | igcewieling | The FreePBX distro. |
| 16:21.07 | Kobaz | oh, no we don't use freepbx |
| 17:15.49 | igcewieling | I have a mix of FreePBX and non-FreePBX boxes. Working on plain Asterisk is much more interesting. |
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| 17:31.35 | drmessano | 12:10:28 <igcewieling> Kobaz: if you consider all the time you've spent on this, you could have purchased an inexpensive linksys/sipura/whaevertheyarecalledtoday instead and still come out ahead. <-- LOL |
| 17:32.02 | drmessano | We've got 10k users and softphones have been indispensable |
| 17:32.22 | drmessano | I hate to say this, for the boomers in the room |
| 17:32.35 | drmessano | But AoIP apps are the future |
| 17:32.38 | drmessano | and the present |
| 17:32.50 | drmessano | There is no Microsoft Teams desk phone |
| 17:33.18 | drmessano | Nobody is buying out warehouses of desk phones for WFH |
| 17:33.30 | igcewieling | drmessano: do you also do desktop support? |
| 17:33.47 | drmessano | I do |
| 17:34.29 | igcewieling | I don't. We are an ITSP/ISP/CLEC, not a desktop support company. |
| 17:34.47 | drmessano | I don't support our phones |
| 17:34.59 | igcewieling | We support our hardphones. |
| 17:35.01 | drmessano | Next time you ask a leading question, adjust your aim |
| 17:36.14 | drmessano | However, an end user with a headset is quite capable |
| 17:37.02 | Samot | 1:34:29 PM <igcewieling> I don't. We are an ITSP/ISP/CLEC, not a desktop support company. <-- You host PBX systems in the cloud for your end users, correct? |
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| 17:37.42 | drmessano | Oh and as the goal post moves on quality |
| 17:37.47 | drmessano | Because it always does |
| 17:37.59 | igcewieling | Samot: only about 1/3 of them and they have hardphones. |
| 17:38.20 | Samot | OK but are they getting a hosted PBX that's is 100% on them? |
| 17:38.34 | drmessano | We're running LIVE broadcasts using those same headsets, with SIP, OPUS codec, and you really can't tell much of a difference |
| 17:38.37 | Samot | Or are they getting it as some variant solution to give them UCaaS? |
| 17:39.08 | drmessano | our SIP client is a modified softphone, basically |
| 17:39.21 | igcewieling | Samot: no, we have not done anything with my idea of running individual FreePBX instances. We currently have a Bicom PBXware MT for our Hosted service. |
| 17:39.49 | Samot | OK, so for those end users your answer for the WFH scenario is what? |
| 17:39.55 | Samot | Call Forwarding? |
| 17:40.03 | Samot | Take your phones home and good luck? |
| 17:40.38 | igcewieling | Well, they can take their hardphones home if they get a power supply for it. They can forward to their cellphone (this is the most common) or they can use a Zulu or gloCom softphone. |
| 17:40.58 | drmessano | lol |
| 17:41.01 | igcewieling | I think we are up to something like 20 glocom licenses. |
| 17:41.30 | igcewieling | I doubt most of them have used softphones before. |
| 17:41.46 | drmessano | wow |
| 17:42.00 | drmessano | So you were completely unprepared for this? |
| 17:42.28 | igcewieling | I'm pretty sure more than 1/2 of our numbers are with customers who just want analog handoff to their current PBX. |
| 17:42.36 | Samot | I do have people that took their hardphones home because they can barely use those so Bria would have made their brains explode. |
| 17:42.38 | Samot | I get that. |
| 17:43.37 | drmessano | There's always "some", but those 10% are the 90% of your support calls, which you've already taken into account |
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| 17:45.49 | igcewieling | The vast majority of our users don't even know our support number because all support goes through one or two of the customer employees. |
| 17:46.55 | igcewieling | usually their IT manager or office manager. |
| 17:48.06 | drmessano | So you hate softphones, but since you don't touch endpoints anyway, you're really basing that on personal bias and not any sort of daily interaction |
| 17:48.07 | drmessano | Got it |
| 17:48.56 | igcewieling | We support the hardphones, the end users go through the local person in charge of phones for support. |
| 17:50.15 | igcewieling | that local person sends the request off to our support |
| 17:50.30 | Samot | In my case of being an ITSP/ISP/CLEC or any combo of those over the years, we've always managed the endpoints. |
| 17:50.44 | Samot | BYOD, sure go for it. It's all on you. |
| 17:51.00 | Samot | Fully configured and interop'd phone to services, managed phone from us. |
| 17:51.13 | igcewieling | We also manage the switches we install which the phones are connected to an the router we install |
| 17:51.50 | drmessano | So please explain your vast experience with softphones |
| 17:55.18 | igcewieling | Every one I've tried sucked. Every one we've deployed to customers was no longer used in 6 months. For a while Sales tried pushing softphones, but went back to pushing hardphones pretty quickly. It has been a couple of years since I've done much with softphones. |
| 17:57.20 | Samot | Sucked how? |
| 17:57.33 | igcewieling | Since the start of the crisis we have had a lot more interest in softphones and have deployed a number of them to customers. |
| 17:58.17 | igcewieling | Samot: usually audio quality and dropped registrations |
| 17:58.33 | Samot | So basically the same issues a hardphone over the Internet would have. |
| 17:58.44 | Samot | All of which can be mitigated. |
| 17:58.59 | igcewieling | not when I use a hardphone on the same internet connection |
| 17:59.22 | Samot | Outside of the fact it's a standalone device. |
| 17:59.38 | Samot | Not running as an app on something else that could be tweaked or modified. |
| 17:59.55 | Samot | Push services, Proxies, all these are able to help mitigate these issues. |
| 18:00.03 | igcewieling | Correct, not running an app that needs to be tweaked or modified. |
| 18:00.18 | igcewieling | 8-| |
| 18:00.57 | igcewieling | In any case, I'm sure you have plenty of advice for Kobaz |
| 18:01.20 | drmessano | Dropped registrations? |
| 18:01.43 | drmessano | So the computer is unstable |
| 18:03.07 | drmessano | I would think if the computer is too unstable to support keeping a softphone registered and has sound issues, then it's not stable enough for them to actually do other work |
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| 18:22.37 | Kobaz | Samot: all of which? |
| 18:22.43 | Kobaz | Samot: 99% packet loss can be mitigated? |
| 18:23.03 | crandon | Hi, I'm a total newbie to asterisk. I managed to set it up to handle sip calls between 3 peers (gate, usera, userb). They can call each other with video and this works fine. Now, I created and extension 'family' like this: exten => family,1,Dial(SIP/usera&SIP/userb,10) |
| 18:23.08 | Samot | No, audio issues and dropped registrations can be. |
| 18:23.17 | Samot | I didn't say anything about packet loss. |
| 18:23.23 | Kobaz | I believe that falls under audio issues, heh |
| 18:23.25 | Samot | Or poor internet connections. |
| 18:23.33 | Samot | So does NAT |
| 18:23.37 | Kobaz | Just send max bitrate opus and pray |
| 18:23.42 | crandon | Whet gate calls family, both phones ring and if any picks up, the call is estabilished, but only audio works, video does not any ideas? |
| 18:24.17 | Kobaz | crandon: are both ends offering a common video codec? is video enabled in the endpoints? |
| 18:24.27 | Kobaz | !debug |
| 18:24.32 | Kobaz | ~debug |
| 18:24.32 | infobot | ACTION DeBuggers $1 |
| 18:24.38 | Kobaz | isnt there a bot thing |
| 18:24.50 | Kobaz | it's been a while... but anyway |
| 18:25.05 | Kobaz | crandon: dpaste.com sip debugs, and wireshark pcap preferable |
| 18:25.14 | Kobaz | and/or tcpdump. doesn' |
| 18:25.18 | Kobaz | t matter, either one |
| 18:26.00 | crandon | Kobaz: yes. Otherwise even peer to peer would not work. BTW: should there be a 'family' peer set up with video enabled as well? |
| 18:26.34 | Kobaz | crandon: not necessarily, depends on the endpoints |
| 18:26.48 | Kobaz | possibly, i could imagine certain endpoints could transcode |
| 18:27.19 | Kobaz | crandon: 'peer' so are you using chan_sip? have you collected your sip debugs yet? |
| 18:28.01 | Kobaz | speaking of video. i've noticed that vp8 through asterisk over wifi has really bad quality |
| 18:28.04 | Samot | The call is going to be established. It is then up to the receiving endpoint of that call to tell it to use video. |
| 18:28.16 | Samot | So there's going to be a re-INVITE that updates the SDP. |
| 18:29.52 | drmessano | 14:23:38 <Kobaz> Just send max bitrate opus and pray <-- Doing that already.. Works great |
| 18:30.05 | crandon | Kobaz: I'm googling how to do it right now, and testing how adding the family peer changes the situation |
| 18:30.11 | Kobaz | drmessano: hah nice |
| 18:30.15 | drmessano | 14:03:08 <drmessano> I would think if the computer is too unstable to support keeping a softphone registered and has sound issues, then it's not stable enough for them to actually do other work |
| 18:30.45 | Kobaz | drmessano: i remember being thoroughly impressed when digium did the 50% packet loss demo at astricon |
| 18:31.04 | Kobaz | drmessano: people work in some pretty unstable environments |
| 18:31.05 | drmessano | All of these nighmare scenarios with softphones are predicated on issues that would otherwise make all of their other work impossible |
| 18:31.30 | Kobaz | i've been roped into cleaning out a 100+ malware infestation because they couldn't actually download the soft phone app because the browser kept crashing |
| 18:31.42 | Kobaz | but they were otherwise working 'just fine' |
| 18:32.18 | Samot | I had someone complain audio was chopping. |
| 18:32.23 | drmessano | Is "just fine" their assessment or their actual productivity? |
| 18:32.24 | Samot | I had someone complain audio was choppy. |
| 18:32.30 | crandon | linphonec reports Media streams established... (video) regardless if I add a family peer to sip.conf |
| 18:32.41 | Kobaz | linphone, fun |
| 18:32.44 | Kobaz | join the club of issues |
| 18:32.47 | Samot | Turns out numerous systems were infected and not only was the audio choppy, normal Internet sucked. |
| 18:32.59 | Kobaz | linphone's tcp/tls handling seems wanting |
| 18:33.11 | Kobaz | i'm having issues with one out of every 15 calls just timing out |
| 18:33.26 | crandon | Kobaz: oh yeag, I've been strugling with it for 2 days now. Now at least the p2p call works between android linphone and linux client. |
| 18:33.30 | Kobaz | and then if the user leaves their phone running for a few hours, they wont receive calls |
| 18:33.34 | drmessano | Because a hard phone is only going to be marginally better under 90% of these arguments |
| 18:33.36 | Kobaz | crandon: try MicroSIP |
| 18:34.05 | Kobaz | i like linphone as a concept, but i need to download the source and start fixing shit |
| 18:34.20 | crandon | Kobaz: I need a cli client or some simple sdk for one end as it will be a 'doorbell' |
| 18:34.28 | Kobaz | pjsip cli client |
| 18:34.38 | Kobaz | sdk... use asterisk as a UA |
| 18:35.49 | crandon | Kobaz: but wouldn't that would require to run asterisk on the 'doorbell' or expose the camera/microphone some way, so that asterisk could access it? |
| 18:36.03 | Kobaz | i dunno |
| 18:36.09 | Kobaz | i have no idea how you have this set up |
| 18:36.55 | drmessano | What's the difference between running Asterisk on the "doorbell" or a cli app/some other SDK? |
| 18:37.04 | crandon | Kobaz: I haven't, but the idea was to have a RPI at the gate, asterisk at home, and android devices as the in/oo-home devices. |
| 18:37.31 | drmessano | So put Asterisk on the RPI as the interface to the gate |
| 18:38.54 | Kobaz | yup |
| 18:39.04 | Kobaz | does a dual ethernet pi exist yet |
| 18:39.48 | Kobaz | http://www.industrialberry.com/ethernetberry-v-1-1/ |
| 18:39.52 | Kobaz | i guess you need to add on |
| 18:39.58 | crandon | drmessano: that's how I understood one of the options proposed by Kobaz, but the reason I'm hesitant to do so is, that it's connected to an untrusted network and connects via VPN home, so in order to let the Android devices register to it I'd need to get them through my home router, to the vpn server to the raspberry |
| 18:40.18 | drmessano | You don't need to let them register to it |
| 18:40.38 | crandon | drmessano: how would I be able to notify them? |
| 18:40.38 | Kobaz | you can firewall the crap out of it |
| 18:40.40 | drmessano | Asterisk is a telephony toolkit, not just some SIP registrar/PBX |
| 18:40.54 | drmessano | You can run Asterisk on the RPI and inside if you want |
| 18:41.07 | drmessano | You asked for a CLI/SDK for the doorbell |
| 18:41.08 | Kobaz | yup, i got asterisk running on the pi once |
| 18:41.13 | drmessano | Asterisk suits this role |
| 18:41.32 | crandon | drmessano: ah you mean to use a 3rd party registrar instead of using asterisk as one? |
| 18:41.40 | drmessano | Dude no |
| 18:42.26 | drmessano | Asterisk+RPI for the doorbell, then whatever you planned to run inside for the Android devices |
| 18:43.12 | drmessano | You asked for a CLI/SDK for the doorbell <-- The answer to that one was *Asterisk* |
| 18:43.24 | drmessano | You said you planned to put Asterisk in the house for the Android Clients |
| 18:43.27 | drmessano | So still do that? |
| 18:43.28 | crandon | drmessano: ah ok, so even 1 asterisk as a UA on the raspberry and 1 asterisk as PBX/registrar on the VPN server, or wherever. |
| 18:44.03 | drmessano | Yeah, you can run Asterisk in more than one place. It's free |
| 18:44.04 | crandon | (or even use a 3rd party registrar on the internet) |
| 18:46.05 | drmessano | *shrug* |
| 18:46.11 | drmessano | Why not just stick to Asterisk |
| 18:46.27 | crandon | The reason I'd like to use asterisk (as a B2BUA) is that I'd like to be able to call a group of accounts, which any could answer the call. So the question is, if the current situation is related to the client (linphonec) in which case using asterisk on the PI could be a solution, or it's related to my PBX/SIP registrar config, in which case it wont help |
| 18:47.01 | drmessano | Linphone is always a problem |
| 18:50.17 | Kobaz | the linux linphone works pretty well actually |
| 18:50.29 | Kobaz | the new version of linphone is supposed to include g729 by default as well |
| 18:51.10 | crandon | Ok, I added a logger line, increased verbosity and turned on sip debugging. Captured all port 5060 traffic on the asterisk machine. Anything else? |
| 18:52.11 | Kobaz | pretty good for now |
| 18:53.57 | crandon | and the configs: extensions.conf, sip.conf |
| 18:55.54 | crandon | here you go: https://drive.google.com/open?id=1YvXshge2oZzjhQHx4cKXX8m_b7ejDsIM |
| 18:55.54 | crandon | I was lazy to modify names. So: usera=loci, userb=kata, family=nemeth |
| 18:56.07 | *** join/#asterisk retentiveboy (~retentive@c-73-43-121-243.hsd1.ga.comcast.net) |
| 18:56.22 | crandon | gate is gate I didn't lie about that one :) |
| 18:57.07 | retentiveboy | https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard says an identify object is created and a match is added for each remote host but I'm not seeing those objects with `pjsip show identifies` |
| 18:57.28 | retentiveboy | what else controls creation of those object? |
| 18:57.45 | crandon | All config modification are pretty much at the bottom of both files |
| 18:58.55 | crandon | And the log contains: gate -> nemeth, which is dispatched to loci and kata, and eventually answered by loci |
| 18:59.02 | retentiveboy | duh, rtfm.... nevermind. |
| 19:00.35 | crandon | gate = linphonec, loci,kata = linphone for android, video codec is set to VP8 (as that's the only video codec the linux linphone supports (h264 would be significanlty better)) |
| 19:02.40 | *** join/#asterisk led_belly (led_belly@gateway/vpn/protonvpn/ledbelly/x-06903193) |
| 19:03.27 | crandon | all peers are on the same lan: asterisk=192.168.0.42, gate=0.4... |
| 19:07.49 | *** join/#asterisk derPlexus (~plexus@ip-178-203-131-93.hsi10.unitymediagroup.de) |
| 19:07.54 | *** join/#asterisk guerby (~guerby@april/board/guerby) |
| 19:12.03 | *** join/#asterisk deavmi (~deavmi@165.0.49.28) |
| 19:12.28 | crandon | Kobaz, drmessano: did you manage to take a look at it? |
| 19:14.01 | drmessano | I'm getting prompted to download from google drive, so no |
| 19:14.44 | crandon | Err, how would you like to get it? |
| 19:15.00 | drmessano | Pastebin works fine |
| 19:17.27 | crandon | sip.conf: http://dpaste.com/2FC7SR6 extensions.conf: http://dpaste.com/3BWHQWZ |
| 19:19.31 | drmessano | An allow without a disallow doesn't make sense |
| 19:21.31 | drmessano | So if you need to force vp8, I you should disallow others or disallow all and allow vp8 and your preferred audio codec |
| 19:22.01 | crandon | logs: http://dpaste.com/38582M0 |
| 19:22.17 | drmessano | so do disallow=all then allow=vp8,opus |
| 19:23.45 | Samot | allow is ignored if there is no disallow |
| 19:24.00 | crandon | No change. |
| 19:24.14 | crandon | I mean I've changed it, but the situation is still the same. |
| 19:24.32 | drmessano | Show a log after you've changed it |
| 19:25.09 | drmessano | Your SDP's are a mess |
| 19:26.10 | Samot | Also turn off debug |
| 19:26.15 | Samot | That is totally not needed |
| 19:26.20 | Samot | core set debug 0 |
| 19:26.28 | Kobaz | heh yeah, i said sip debug. not core debug |
| 19:29.29 | crandon | New logs: https://pastebin.com/usMBdRNs |
| 19:30.06 | crandon | Ok, I turned core debug off now. |
| 19:30.25 | drmessano | Yeah post a new call npw |
| 19:30.30 | drmessano | Not reading that other mess |
| 19:32.19 | crandon | https://pastebin.com/PNKP0QLi |
| 19:33.50 | crandon | Now it seems, that the initiating end is not hanging up, when it is initiated from the called party. I don't remember seeing this before. Also I think this is new as well: |
| 19:33.50 | crandon | 2020-04-06 21:31:39:812 ortp-error-Failed to parse SDP message. |
| 19:34.07 | crandon | This is from linphonc (gate) |
| 19:34.28 | crandon | Sorry, got to go now, will keep this open and check back in an hour our so. |
| 20:07.06 | crandon | I'm back. |
| 20:07.15 | crandon | Any ideas? |
| 20:09.37 | Samot | I still don't see a sip debug |
| 20:09.47 | Samot | That doesn't contain "core debug" output. |
| 20:11.16 | crandon | The last one should be it. At least I ran core set debug 0, reload and re-ran the call |
| 20:13.13 | Samot | It's not. |
| 20:13.18 | Samot | It starts with debug output. |
| 20:14.27 | Kobaz | heh |
| 20:14.29 | Kobaz | kill it! |
| 20:21.27 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 20:28.24 | crandon | New one now copied from the console: https://pastebin.com/sX6G2mZ6 |
| 20:28.44 | crandon | (after restarting asterisk via systemd) |
| 20:33.07 | crandon | This is from logfile and contains only sip related info (as far as I can tell): https://pastebin.com/A0sab3Xk |
| 20:38.25 | *** join/#asterisk Micc (~Micc@c-24-18-201-121.hsd1.wa.comcast.net) |
| 20:39.12 | Micc | Is this a good place to ask questions about sipml5 and asterisk webrtc setup? |
| 20:39.22 | *** join/#asterisk salviadud (~rgonzalez@189.207.212.148) |
| 20:40.56 | Samot | [Apr 6 22:25:12] WARNING[21313][C-00000001]: channel.c:5579 set_format: Unable to find a codec translation path: (vp8) -> (opus) |
| 20:40.56 | Samot | [Apr 6 22:25:12] WARNING[21313][C-00000001]: channel.c:5579 set_format: Unable to find a codec translation path: (opus) -> (vp8) |
| 20:41.51 | Micc | Is SIPML5 even a good library to use for WebRTC now? Is CyberMegaPhone better as a starting point for getting WebRTC/Asterisk working? |
| 20:42.21 | *** join/#asterisk deavmi (~deavmi@165.0.49.28) |
| 20:42.45 | Micc | I have the SIPML5 live demo working, but it's kinda screwy. It takes 5-10 seconds before it even sends the invite to asterisk. Not sure if that's a configuration issue or aa SIPML5 issue. |
| 20:45.29 | Kobaz | Micc: dump a capture locally |
| 20:45.38 | Kobaz | see if sipml is actually sending anything when you think it is |
| 20:45.50 | Micc | good idea. |
| 20:45.52 | Kobaz | if you dont see the packets, it's not sending them |
| 20:46.36 | Micc | I was trying to step through the code. I suspect it's doing some ICE or TURN thing. |
| 20:46.51 | Micc | wireshark should be easy to verify that theory. |
| 20:46.58 | crandon | with disable=all and vp8,opus I don't even have sound in p2p connection. |
| 20:49.00 | Micc | crandon, I noticed sound issues as well, but I suspect it might be using a different sound device. For example I believe it's picking a different microphone. |
| 20:49.04 | crandon | Samot: I'm sorry, but this is not saying me anything beside, my p2p test result and the above are likely connected. So far the only combination that provides both audio and video is not setting disallow all _and_ setting allow=vp8. (without setting allow=vp8 I have sound, but no video) |
| 20:49.51 | Micc | crandon, sorry I'm new to the conversation. ignore what I said. sounds like you've got something else going on. |
| 20:49.57 | Samot | You need to allow an audio codec as weel |
| 20:49.59 | Samot | well |
| 20:50.12 | Samot | allow=opus&vp8 |
| 20:50.20 | Kobaz | , |
| 20:50.26 | Kobaz | or does & work too? |
| 20:50.36 | crandon | vp8/opus is not giving audio (allow=vp8,opus as suggested earlier) |
| 20:51.30 | *** join/#asterisk Madomokhtar (~Madomokht@197.246.42.114) |
| 20:51.35 | Madomokhtar | hello |
| 20:51.38 | Madomokhtar | need help |
| 20:52.12 | *** part/#asterisk Madomokhtar (~Madomokht@197.246.42.114) |
| 20:52.15 | Micc | I figured out my problem. It was trying to lookup ICE servers. In expert mode set to [] and it is instant fast. |
| 20:52.25 | Kobaz | nice |
| 20:53.19 | Micc | I'm about to see if my codecs give me working audio. |
| 20:54.17 | Samot | Honestly, off the top my head I can't remember. |
| 20:55.13 | crandon | Samot: so allow=vp8,opus leads to video but no audio (with disable=all) |
| 20:55.56 | crandon | This is the combination that was used to produce the pastbin logs |
| 20:57.57 | Samot | OK then try something that isn't opus. |
| 20:58.05 | Samot | like ulaw/alaw |
| 20:59.10 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 20:59.12 | Micc | I have partial audio. My main machine doesn't seem to be playing anything from my laptop, but I can hear everything on my laptop from my main machine. So I'm guessing my desktop is trying to play the audio out the wrong device or something. |
| 20:59.35 | Micc | I'm using allow=opus,vp8,vp9,h264 |
| 21:03.25 | crandon | Samot: Unable to find a codec translation path: (vp8) -> (ulaw) same for alaw |
| 21:04.22 | Micc | when I do rtp set debug on I can see packets being sent that say "(via ICE)" but I don't think I have an ice server unless asterisk is one by default? |
| 21:04.59 | crandon | Samot: pcm works. |
| 21:05.14 | Micc | I see a lot of these messages filling my screen too: SRTP protect: replay check failed (index too old) |
| 21:05.27 | Micc | I suspect that's unrelated though. |
| 21:07.28 | crandon | Ok, so disallow=all allow=vp8 allow=pcm result in a working p2p audio and video call, however calling the 'group' extension and picking it up on the same phone as before results in sound only and no video |
| 21:07.42 | crandon | Would a new sip debug output help at this point? |
| 21:10.15 | Kobaz | always |
| 21:11.17 | Micc | Does it not work on Microsoft Edge? I get an error about createPeerConnection |
| 21:12.09 | crandon | This is driving me now not even peer call works with pcm... |
| 21:19.58 | crandon | If i'm omitting any codec configuration vp8 and ulaw is selected by the peers (as reported by phone client), but setting allow=vp8,ulaw results in channel.c:5579 set_format: Unable to find a codec translation path: (ulaw) -> (vp8) |
| 21:23.27 | *** join/#asterisk deavmi (~deavmi@165.0.49.28) |
| 21:24.31 | Micc | I can only get one side to work at a time. I feel like this is a bug with sipml5 but I can't be sure. Only the one who places the call can hear the other person. Which ever device answers the call does not get audio but the microphone works. |
| 21:27.14 | Micc | audio only call works, but when I do video only the one placing the call can hear audio. |
| 21:31.18 | Micc | does the 'simple-bridge' basic-bridge not support audio and video? |
| 21:32.22 | Micc | oh a video call I can see they both join the bridge twice. Only once for each with audio call. |
| 21:33.12 | Micc | That makes sense. Maybe I should try other audio codecs. |
| 21:34.20 | Micc | How can I see which codecs a call is using? I used to be able to do sip show channels is there a similar command for pjsip? |
| 21:36.13 | Micc | I see pjsip show channels but I can't tell what codec is being used. |
| 21:36.17 | *** join/#asterisk chandoo (~chandoo@pool-100-1-166-161.nwrknj.fios.verizon.net) |
| 21:38.35 | crandon | Ok, so it seem the working combination with disallow all is to have alaw first and only after that vp8 |
| 21:38.49 | crandon | at least for the p2p call. |
| 21:39.42 | crandon | But video is just not working with 'group call'. |
| 21:46.16 | *** join/#asterisk CarlosTico (~CarlosTic@107-134-203-65.lightspeed.wlfrct.sbcglobal.net) |
| 22:08.48 | Samot | How are you doing this group call? |
| 22:18.48 | *** join/#asterisk matrix1233 (~matrix123@185.61.186.147) |
| 23:19.29 | drmessano | It's been a few hours.. do all softphones still suck? |
| 23:23.56 | Micc | drmessano, lol, We've been using SessionCloud, it had a lot of bugs, but it's gotten a little less suck. The QR Code provisioning is pretty slick. |
| 23:24.27 | drmessano | ok |
| 23:24.38 | Micc | I'm trying to find out if all webrtc with asterisk sucks. |
| 23:24.44 | Micc | and it seems it does suck. |
| 23:25.46 | drmessano | Well, don't ask in here |
| 23:26.00 | Samot | Hrm. |
| 23:26.00 | drmessano | You'll get a few good answers from the developers, sure |
| 23:26.05 | Samot | Here we go again. |
| 23:26.05 | file | it works fine, we use it every day and Sangoma products use it |
| 23:26.07 | Micc | Is there a better channel for that? |
| 23:26.08 | Samot | ^^^^ |
| 23:26.12 | Samot | WebRTC works fine. |
| 23:26.24 | file | it's just not "plug and play" and SipML5 is not something we use, I've had better experience with JsSIP |
| 23:26.24 | Samot | It is used by numerous people. |
| 23:27.11 | Samot | This is getting into the realm of logic that IPv6 or VPNs must be used to solve NAT issues. |
| 23:27.14 | drmessano | Stick to chan_sip, hardphones, and PRI's if you want stability. AKA the B.O.O.M.E.R. Stack |
| 23:27.16 | Micc | yeah, appr.tc/ seems to work fine with webrtc, but nothing seems to work 100% with asterisk. |
| 23:27.33 | Samot | Micc: That's because Asterisk is a toolkit. |
| 23:27.39 | Samot | By default, it's a blank slate. |
| 23:27.46 | Samot | YOU have to make it do what YOU want. |
| 23:28.22 | file | the Wazo guys also are using WebRTC with Asterisk for video conferencing and such |
| 23:28.27 | drmessano | Oh and only on bare metal, because VMs are kind new |
| 23:28.34 | Samot | I know it's always referred to as a "Software PBX" because that's the primary use case for it but it's not. |
| 23:28.43 | Micc | I just need one good starting point and I'll get there. But both the tutorials on the wiki leave some things out. |
| 23:31.01 | Micc | What channel would be best for help with WebRTC? I need to get something setup that will work with modern browsers on any platform, windows/mac/linux/android |
| 23:33.52 | file | I don't have an answer for that because there's tons of moving parts and it depends... |
| 23:34.15 | file | WebRTC is just layers of technology in a browser combined with a Javascript API to actually use it |
| 23:34.34 | file | or a library from Google which can be used in native applications |
| 23:34.55 | file | it doesn't even define a signaling layer, like SIP, so there's libraries that add that and help out - such as JsSIP |
| 23:34.59 | file | some of which include examples |
| 23:35.26 | file | then there's the Asterisk side, which is kind of independent in a way but has its own configuration |
| 23:35.58 | file | it's all just ways to build things |
| 23:42.51 | *** join/#asterisk matrix1233 (~matrix123@185.61.186.152) |
| 23:45.53 | *** join/#asterisk led_belly (led_belly@gateway/vpn/protonvpn/ledbelly/x-06903193) |