I have been working on implementing a Sipml5 phone for a clients website for the past week or so. I had a asterisk 1.8 system running on the clients end (current config w/ zoiper and hard phones, etc). Their new system will use a webphone (thats the plan). SO far putting Sipml5 together has proved more difficult than I initially thought.
The actual tutorial for it is completely useless, and I messed around with multiple asterisk installs until I finally got it working as far as registering, etc… But the Sound was only 1 way, different version is one way and the other way is static, etc…
I finally got the webrtc2sip server installed aswell but there is basically 0 documentation on how to configure it properly.
Today after reading on this forum that it worked with 11.5 I installed 11.5 (I had 11.5 before but it wasnt working) and using the configs I made over the past days it works calling from chrome to chrome or chrome to chromium. But I cannot call a regular SIP client and the asterisk server keeps crashing after every call ends, or in the call and there are no errors, not even in debug, it just says asterisk ending clean(0).
Any ideas on this? Any help would be greatly appreciated! Thanks in Advance!
I recommend to use the gateway WebRTC2SIP with asterisk, I have asterisk 1.8.X and Asterisk 11.2, 11.4 working with it. The technical guide is a great help to configure the media gateway so take a deep read on that document and try again.
Well where is the guide? Because I have the “Technical Guide” and its really not very descriptive when it comes to what to do…
Here is the technical guide–>http://webrtc2sip.org/technical-guide-1.0.pdf
Dude all the information are inside that PDF if you can’t understand how to configure it then you need to read a lot about protocols, codecs and the sipml5 background and connectivity.
If you are looking for step by step information then I recommend you to move the thread to the Job forum.
On the other hand if you still have issues understanding how to use the media gateway then try the direct connection between sipml5 and Asterisk. The asterisk wiki has an article on how to configure that.
Yes I got it setup directly through asterisk and I did follow the technical guide but it doesnt seem to connect to my asterisk server. It only works from chrome to chome and fails to call a normal SIP endpoint due to SRTP apparently…
Ah yes, if you use only ‘WS’ seems like those issues disappear and you can call normally.
So you are saying I have to try to force my SIP clients onto WS aswell? But most of them don’t seem to support it…
I managed to fix it by editing the SIPML5 code. It was hardcoded to proxy through their server. Changing the actual code to make it check properly and pass through our server 100% allowed it to work fine using the new asterisk 11.5.
I have an issue with the audio not coming back after taking the phone off of hold. When Hold is pushed in the SIP client.