WebRTC Browser Phone with Asterisk & Raspberry Pi - Part 2 (PJSIP)

This is the next part in the the two part video on Installing a Browsers Phone with Asterisk and Raspberry Pi. In this video I will show you how to complete this with PJSIP as the channel driver. The results will be exactly the same, however we do have to use Asterisk 16.
Note: You must first complete the first video:
• WebRTC Browser Phone w...
(The above video covers setup and Certificate generation. Its based on Asterisk 13 with chan_sip)
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[@diggybell]

This is another excellent tutorial with the addition of PJSIP. Just this morning I made the decision to migrate our production hosts from CHAN_SIP to PJSIP. Your well organized and clear configuration examples will be essential to get this running as quickly as possible. Kudos once again for your effort, clarity, and presentation style. You are to be commended for a perfect balance if standard speaking, code/configuration examples, and clear demonstrations. Most KZbin technical content creators could benefit from your presentation style.

[@InnovateAsterisk]

From Asterisk 16 and higher, it's really better to use PJSIP, but in 13 and lower, there are still some bugs. Most of my Asterisk work is now on AST16 with PJSIP - I'm also using Realtime engine with PJSIP, although I found their approach to the database management and structure a little too abstracted - I like to know how stuff works... while they take the approach, "just run this". Anyway, thanks for the feedback and good luck!

[@tamerlannusraddinov]

The best content on asterisk. Thanks

[@InnovateAsterisk]

Glad you liked it! More on its way, stay tuned.

[@IronMan-nt7gu]

First off, excellent video on the subject, the concepts were explained very clearly. I'm new to sip and asterisk, so exploring/experimenting based off your videos. Currently I have asterisk configurations as below, 6002 is a webrtc client and 6001 is normal sip client (linphone). My webrtc client is an application i coded using the JSSIP library. I'm testing out in local LAN network. However, i'm facing issues as below when I tested a video call. 1. sip client -> webrtc client : video/audio works perfectly fine. can hear and see both sides 2. webrtc client -> webrtc client : video appears but audio can only be heard in one direction 3. when attempting a blind call transfer in webrtc client, the sip debug messages state 'Contact header not specified' I'm confused whether this issues are due to my asterisk configuration or jssip library. I checked out another library sip.js but its in typescript language which I'm not familiar with. I'm comfortable with C, javascript & python. Please guide recommend any tips, debug steps or libraries i should try out. [6002] type=endpoint context=default webrtc=yes direct_media=no ;remove_existing=yes aors=6002 auth=6002 allow=opus,ulaw,vp9,vp8,h264 use_avpf=yes media_encryption=dtls ice_support=yes media_use_received_transport=yes rtcp_mux=yes max_audio_streams=10 max_video_streams=10 dtls_auto_generate_cert=yes [6001] type=endpoint context=default allow=ulaw,alaw,g722,gsm,vp9,vp8,h264 auth=6001 aors=6001 max_audio_streams=10 max_video_streams=10 dtls_auto_generate_cert=yes

[@InnovateAsterisk]

Thanks for the feedback. Please try out the browser-phone at: www.innovateasterisk.com/phone/ and if the same issues exists, then it's jssip, or your own code. All the best!

[@josielvsilva]

Very very good is content! I’m from Brazil. It helped me moch. Congratulations!

[@InnovateAsterisk]

Awesome! Thank you!

[@MartinLujan5]

Great Job!! from Argentina

[@InnovateAsterisk]

Thank you!

[@rubenjerez4282]

Gracias desde Mexico, increible video

[@InnovateAsterisk]

Glad you like it!

[@ronnytjoa2420]

Thank you for great tutorials. Can you please add a tutorial on XMPP features with Openfire?

[@InnovateAsterisk]

Absolutely! Coming soon.

[@noway0815]

very usefull, I will see it again and again till I setup my pgsip... But could you please make a video for pjsip vs sip in details... thank you. big like

[@InnovateAsterisk]

Thanks for the feedback! Great idea, will do.

[@noway0815]

@@InnovateAsterisk thanks, graet, hope soon

[@mikes6672]

If there was a medal for it all three would have to go to you

[@InnovateAsterisk]

Thanks ;)

[@jasonshen]

very nice video, it would be better if you provide the finished files, as typos are a nightmare trying to follow, but great work cheers!

[@InnovateAsterisk]

Thanks for the feedback - I will be using GitHub more in the future videos for further information and to be able to copy and paste the commands.

[@jasonshen]

@@InnovateAsterisk can you upload the final files for this tuts, as i have issues getting it working, i think due to typos cheers

[@SunnyKhetarpal]

This is amazing ....Thanks

[@InnovateAsterisk]

Glad you like it!

[@Nivedh96]

Nice one...👍

[@InnovateAsterisk]

Thanks ✌️

[@novebmer11]

It is awsome bor, Thanks.

[@InnovateAsterisk]

Glad you like it

[@jesucristoesteban908]

Keep making videos llike this, please. Do you have a Patreon or any other support channel? I would like to be part of something like this in any way. Really really thank you for all your effort, i cant wait to share it with my friends. Thank you!

[@InnovateAsterisk]

It's my absolute pleasure to make these videos! I have unfortunately has no time since COVID threw a spanner in the works, but I have made a plan for next year and will be starting again in the 2021... stay tuned, many more videos coming!

[@jesucristoesteban908]

@@InnovateAsterisk The pandemic has been difficult for many of us, in many ways, but it's our work to keep the wheel of creativity running and you are dealing an amazing work with this content. Thanks for all again!

[@avinashbabanagare59]

its awesome. Can we have feature to dial mobile numbers also without using buddy feature ? Can we use this in intranet (without internet in local network) ?

[@InnovateAsterisk]

This project allows you to make any call, local or mobile but it requires that you own PBX is configured to do so. This code only replaces the need for a hardware phone like a Yealink or mobile phone application like Zoiper.

[@sunterthinkpad]

It's worked. Is there possible to modified a SIP can sending a file also?

[@InnovateAsterisk]

I'm working on this. Coming soon, can't say when.

[@jeanrenoir95]

another question: i will try to run it over internet. What NAT or port forwarding must be enabled outside the 80, 443and 5060 ?

[@InnovateAsterisk]

Lovely question! I will do a full video on this, but basically you only need the port(s) you specified in the http.conf file. So, probably TCP:443 or 4443. Also check out rtp.conf there is a UDP port range for media (the RTP streams). Most importantly DO NOT open port UDP:5060

[@Carlosknoche]

hello, thank you for your video, i got a doubt: is possible use chan sip an pjsip at the same time in this projects?

[@InnovateAsterisk]

Asterisk has issues if these two modules are loaded, especially around handing webrtc requests... it and either-or situation.

[@santiagozambrana2147]

Excellent video!!, have you tried to create a dashboard with Node-red for making calls??

[@InnovateAsterisk]

Thanks! I have not looked into Node-red... give it a try :)

[@kcyeohmy]

Just want to ask if raspberry pi have any FXO and FXS connectivity? RJ11 cable. using asterisk. let say if i have a one landline. can i use the port to plug into raspberry pi and then add some extensions? I would like to have at least 5 to 10 concurrent call. like a mini call center. and is it possible for these numbers to be added via whatsapp or wechat so that we can let customer call from that Apps?

[@InnovateAsterisk]

Thanks for this fantastic question. The answer is this: There are various way to connect a Raspberry Pi with a traditional analog line, and I have tested one way. I have used a Grandstream SIP ATA, it comes with (1x FXS 1x FXO), and the way it will work is that you register this separate device on your network (Connected to your local switch) and then use it like a gateway, so you trunk dial out over this network. It will only have one line, so you would need to have your fixed line operator hunt for an available line if you buy and setup say 4 or 5 lines. (Outbound would also be done the same way.) I have seen something like this using USB, but have not tried it. I would assume it would list then use USB drivers rather than a IP network connection. If you are willing to spend a bit more, you could look at SIP gateway. They are normally starting at 4 port, and then go up to about16 port (more than that you would probably be using BRI/PRI technology). Either way, you would terminate call onto this hardware, its converted to SIP, and then forwarded onto your Raspberry Pi over your local IP network.

[@djameleddinekhe4035]

hello everyone, is there a development for multi-party video conferencing with BrowserPhone. these are very much in demand in my company.

[@InnovateAsterisk]

The Browser Phone now supports all Asterisk viewing modes in ConfBridge: github.com/InnovateAsterisk/Browser-Phone By default it supports 16 in SFU, other modes don't have limitations. (Only features not implemented are conference chat messages).

[@AjayPattni-jb4uh]

Hello Sir, I need same solution I need to call from html Client side (Angular or Javascript) and call asterisk server. Please share any reference. without tlsbinding Please suggest how to configure asterisk server? and configure javascript.

[@yohannesephrem9096]

I'm just started on wetting my feet on astrisk and the whole open-source telephony business and I very much appreciate the work that you're doing. As a newbie, I want to ask you this, can I deploy the whole astrisk stack with the webRTC feature for video sip to an intranet environment. Thanks in advance.

[@InnovateAsterisk]

Technically yes, it can but it has some drawbacks.... From the Asterisk side, it really can be run "off-line" and doesn't need the internet at all. Just be sure so disable any srv lookups, and its normally better to use everything on IP, in this case nothing would then need to be looked up in a DNS server (they only really work on the internet). With the Browser Phone, it can me used off-line, but all WebRTC calls have to use ICE/STUN to establish a call. This lookup can take 500ms to timeout/fail - so this can create a slowness in the call setup. Another thing to do will be to download all the files that are typically on the CDN server.

[@veeranjaneyuluboyapati7522]

how to make PSTN calls using this solution, pl. explain

[@InnovateAsterisk]

You will need to trunk in/out (connect with) a voip service provider. Stay tuned, ill be doing a video on this soon.

[@veeranjaneyuluboyapati7522]

​@@InnovateAsterisk Thank you so much, you are doing great.

[@MrTomfernandez]

Thanks for share this! its amazing! I made a spanish language pack for the phone, can I send you? I have a question, why use SipJS 11 and not 18? I try to use 13 but phone.js need modifications, so for 18 version should be a lot of changes to do. Sorry for my English =D, greetings from Argentina!

[@InnovateAsterisk]

Thanks for the feedback, glad to share. For language pack files, its best if its done via GitHub, fork it, and submit your changes. github.com/InnovateAsterisk/Browser-Phone With regards to SIPJS version; I have not had a good look at the change log, but as yet, I'm not able to see anything that is causing an issue. So the upgrade will just create a lot of work, but for no real reason. As soon as I encounter an issue that's preventing something from working ill be sure to upgrade. Something important to remember - most of the heavy lifting is actually done with your browser, and javascript, the SipJS library is just a nicely written wrapper for the PeerConnection object. developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection

[@MrTomfernandez]

​@@InnovateAsterisk thanks for such great work!

[@pramodrajput1234]

Great tutorials. I am working with asterisk 16 on Centos 7 platform. With the help of your video going all things are good but having an issue with the call recording feature. The call recording button working fine but after completion, the recorded call does not work. can you please help?

[@InnovateAsterisk]

Someone has mentioned this before. There may be a bug, but, ill double check. The solution was to remove the CallRecording Database (Index DB) using Developer Tools, and then record a call again. It seems that for some reason the database was made, but the table structure is either not made, or not made in time, leaving the database in an unusable state.

[@pramodrajput1234]

@@InnovateAsterisk Thanks, It's working now

[@tomascruz2422]

awesome solution, i tested it and the features works fantastic. but only few things still missed or i am doing wrong, every time after connected logged, contact information is lost. and manually need to add contact.

[@InnovateAsterisk]

Glad you like it! :) Well, the way this solution works is that it will save buddies (or contacts) to your Browsers local storage engine. If this data is deleted then you will have to add it again. It may be from an application that you have installed, or the way you have setup your browser, that this information doesn't persist across browser sessions. Raise this issue on the GitHub issues page, so we can discuss further. github.com/InnovateAsterisk/Browser-Phone/issues

[@tutsw]

Hi! I liked your project very much, cool, but the sound through the GSM gateway does not work for me, there is no OPUS codec, is it because of this? Could you help me?

[@InnovateAsterisk]

If the trunk does not have opus in the allow list, and the extension does, then transcoding will take place (most of the time). It will also depend of a number of other factors but this should work fine. Generally speaking when extension-to-extension works, and trunk-to-extension does not, its to do with the primary difference... mainly NAT!! The GSM trunk is from the "outside", and this involves firewalls etc. Raise this issue here: github.com/InnovateAsterisk/Browser-Phone/issues so that if someone else has the same issue it may help them.

[@ron1729]

Excellent video! I'm looking to build a web app which can make video calls to existing Cisco or Polycom VC devices through sip or h323. It seems like what you have here can do just that? Is that correct? Kind of new to this. Thank you so much for your sharing!

[@InnovateAsterisk]

Thanks for the feedback. Yes, this app will be able to setup a video call, but it will only use SIP, no other protocols are supported. I have not tested this with Cisco or Polycom, but if they follow the SIP protocol - they will work!

[@benwheadon3079]

Hey Conrad, great series! is there a way to email a link to someone so they can join a call from their browser without "registering" them as an extension?

[@InnovateAsterisk]

Thanks for the feedback! You could make the page server driven, and create a temporary random auth (in the DB), that you then push into the settings (client side) with the localDB.setItem(...), this would allow the phone to register as it loads, much the same way as when you refresh/reload the page. You would probably want to then remove the row in the database after that, as this would leave your systems venerable... you just need to view the source code and get the username and password. If this is for a closed/corporate network, then is risk is considerably lower, and you could even auth with IP. Good luck

[@hsmptg]

It is possible to coexist chan_sip and pjsip in asterisk? The vídeo of my doorbell (Dahua VTO2000A) seems to work only with chan_sip!

[@InnovateAsterisk]

It's not recommended and some things like websockets require one or the other, but not both. If you are on. Asterisk 16+ go with pjsip only, otherwise stick to chan_sip. Even tho pjsip has been part of Asterisk for many years now, many years of documentation across many companies still have to change. Personally at this time I would be using Asterisk 16 with pjsip. The doorbell should work over pjsip just the same as over chan_sip.

[@jeanrenoir95]

ok, i just spent all the night on the first video to install v13 and it works fine. Now i am viewing the second video for PJSIP that start with,"hey do it again with v16 this time !". Is there an easy way to upgrade from v13 to v16 without redoing everything ? Thanks for the wonderful job anyway.

[@InnovateAsterisk]

Sorry about that! I suppose you could stay on 13, but it’s quite old now, so will not benefit from any updates. To upgrade, in place, just follow the build steps/commands again, but because the dependancies are already installed, it should take a bit quicker. Otherwise everything stays the same, config files etc. Don’t forget that once you have done the make install command, the executable file will be overwritten, so you must stop and start the service.

[@MinotaurSOH]

Why not do the configuration inside the GUI Dashboard of asterix?

[@InnovateAsterisk]

This channel is about learning the actual config files that go on behind the scenes, so that you know how Asterisk works, and not how the GUI works.

[@jacknahid]

I am getting error : res_pjsip_session.c: 101: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing) No sound in calls . Zoiper response: remote audio : None Zoiper audio : g722 FreePBX 15 asterisk 16 cloud vm ubuntu

[@InnovateAsterisk]

This is often because DTLS is not enabled. WebRTC has to use DTLS as the encryption.

[@thomasknoop1891]

Just wanted to confirm that www.innovateasterisk.com/phone/ works out of the box on a (apparently) correctly configured Asterisk 16.15.0 with FreePBX 15.0.16.81 for voice calls! .

[@InnovateAsterisk]

Nice! Thanks for the feedback, good luck

[@jacknahid]

I install SSL and tryied to configure but i failed . Can you provide me steps for freepbx , please ?

[@techmind7412]

hey bro I want implement in ionic 5

[@InnovateAsterisk]

Great! I understand Ionic is a UI library. I'm sure the Phone will pop in there quite well, I would however suggest that you use an container for the phone. It's very sensitive to size changes and performs multiple operations when doing so. It can work well with responsive layouts, but best as a (from the same domain).

[@brunolongo794]

Hi Conrad, how are you? How can I allow same user to make or answer calls from webphone and from MicroSIP phone? I tried to change pjsip but does not worked. Sample I used: [User3](basic_endpoint,webrtc_endpoint,phone_endpoint)

[@InnovateAsterisk]

There are more videos coming soon that should answer your questions, stay tuned.

[@NoufalKottola1]

I configured as you instructed. Everything working except chat. Unable to find out any error in asterisk cli, but text messages not reaching to endpoint. Kindly advise please

[@InnovateAsterisk]

Start by enabling the SIP trace. On chan_sip it will be "sip set debug on", and chan_pjsip you have to enable history "pjsip set history on", and then pjsip show history. Then on the browser side, make sure the Developer Tools is open, highlight the ws (web socket connection) in the Network tab. On the right, select the Messages Tab. There you can see the SIP messages passing to and from the Asterisk server. There is also a Issues pages on the GitHub site, you are welcome to post support queries there: github.com/InnovateAsterisk/Browser-Phone/issues

[@NoufalKottola1]

@@InnovateAsterisk Thanks , I fixed that issue. that issue due to a SHELL command error. Thank you so much.. You did a wonderful project.

[@pramodrajput1234]

@@InnovateAsterisk Can you guide me? I am working with "pjsip". Text message (sent and received) both show in "Message Tab" under "Network Tab". But received message does not display chat screen. Only sent message display on chat screen. Kindly advise

[@kumark8850]

Hi , Rasberry Pi , hardware is mandatory for this browser phone technology? Can you please clarify

[@InnovateAsterisk]

Thanks for your question. No, not at all, its just that this demonstration is designed to be a start-to-finish solution, that's why the raspberry pi, but if you already have your own Asterisk box (or other compatible SIP server), you can use the browser phone page - it's just a bit of JavaScript. There is more information available at: github.com/InnovateAsterisk/Browser-Phone even an already hosted page: www.innovateasterisk.com/phone/

[@kumark8850]

@@InnovateAsterisk Thank You Very Much for Your reply and Clear explanation. Thanks for sharing the useful technological knowledge.

[@albertoflecha2388]

Awesome vid! Is there a way to integrate WhatsApp business with asterisk?

[@InnovateAsterisk]

Thanks for the feedback. Yes! WhatsApp and more... you would have heard me mention the main Buddy area as the “Stream” the concept behind the stream, as you saw it logs a call detail, then a text message, it doesn’t matter to the system what type it is, so long as it can get a date stamp and a relationship with the buddy/contact. In the pipe line is support for: FAX, Email, SMS, and even social media like twitter and WhatsApp. Stay tuned, coming soon.

[@albertoflecha2388]

Innovate Asterisk That would be awesome. We implemented WhatsApp business over a year ago and we manage it with the web interface, but during this pandemic it has been the most useful tool for connecting with our customers. Since we have a 20 year old phone system we can’t really do much work remote with it other than call forward.

[@InnovateAsterisk]

This pandemic seems to have highlighted the importance of being able to work from home, (most probably) by having a lot of your IT in the cloud. Even if all systems are not cloud based, being able to be fully functional at home, is now the new goal. For a lot of people this can be achieved with internet for others you may need VPN on top of that. It does however mean your telephone line needs to be digital! Asterisk is a popular option to terminate these digital lines. This Browser Phone project is designed to show you that a browser is all you really need to have a fully functional telephone, connected to your Asterisk PBX and digital phone line.