S1E11: 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: https://www.youtube.com/watch?v=mS28vfT8wJ8
(The above video covers setup and Certificate generation. Its based on Asterisk 13 with chan_sip)
Github Project: https://github.com/InnovateAsterisk/Browser-Phone
Your asterisk videos are extremely clear and informative, thank you!
Q: What drove the decision to use Asterisk-13/16 instead of the current Asterisk-19?
Q: Would you please consider researching and sharing the necessary settings to deploy Browser-Phone in Asterisk-19 in AWS on Ubuntu 20.04 using CHAN_PJSIP?
(I’m able to get SIP signaling to work just fine but there is no audio)
Thanks!
Q: What drove the decision to use Asterisk-13/16 instead of the current Asterisk-19?
A: Mainly because there is a large transition from Asterisk 13 to Asterisk 16, especially isn the WebRTC support to do with the chan_sip and chan_pjsip changeover. Up till Asterisk 13, chan_sip appears to work better with webrtc, but then from Asterisk 16, there are features that are better in chan_pjsip. These videos help to assist you to make the choice to change over to Asterisk 16. From Asterisk 16, 17 and up – the setup and modules will all be mostly the same.
Q: Would you please consider researching and sharing the necessary settings to deploy Browser-Phone in Asterisk-19 in AWS on Ubuntu 20.04 using CHAN_PJSIP?
A: I am planning to release a video covering this and more. Stay Tuned!
If there are other technical questions around the Browser Phone, I would recommend to head over to: https://github.com/InnovateAsterisk/Browser-Phone/issues. There is a building community of like-minded individuals that may also be able to help.
I need your help, im using your Browser phone for a personal project where i want show to my boss, and i want to deactivate some other lines where stay available in the ramal.
Because i’m facing some problems, I want him to receive only one call at a time, and the same is receiving more than one.
Causing some conflicts who is receiving, making possible who he wants to answer.
I’m waiting for your answer.
There is an option for call waiting (in settings menu item). Just turn this off (on by default) with this off, you can only receive one call at a time (other calls are rejected with “busy here”).
Also, its best that you use https://github.com/InnovateAsterisk/Browser-Phone/issues in the future.
Hi need your help to setup pjsip i have done all the possible settings as per your video and some database interactions using pjsip, happy to pay for your service, but need your help, appreciate your support.
Hi Sandeep, ill be in a better position to help you in your project next year. I’m busy with a commercial version of the free Browser Phone, called Siperb https://www.siperb.com. I’m busy working on getting the MVP together.
Dear Sir,
I am from Cosmicinfotech, India.
after configured this project from outside network User1 and User2 is connected but voice not come
Please suggest
Regards,
Ghamer
Generally speaking an audio problems is related to NAT. Remember that with WebRTC, the media (voice and video) do not follow the same path as the signalling. The SIP signal travels “in side” the WebSocket connection. This is like a secure point-to-point TLS tunnel. When a call is established there is something called an ICE negotiation – this is where the browser and the server agree on ports and an IP address etc. If this process fails, there may be an established call, but no media, or media in one direction. You would have to inspect the SDP for the outcome of this and possibly use something like wireshark to determine the issue.