Currently we have several PBXs in Asterisk and there is a problem of Flooding of port 8089 Webrtc, it is not an attack since the traffic is valid and comes from IPs of our clients, this problem happens when the person who is using the Webphone has intermittence of internet, generating multiple connections in closing state that are observed in the Kernel log.
We think that to solve this is to use an Kamailio such as Webrtc Gateway and that the flooding is controlled from this point.
We need a professional to help us install and configure an Opensips as a mid-registar WSS, which allows us to log the extensions found in Asterisk of the SIP type.
Observation: The asterisk actualy is not Real Time, and for local devolpment with cannot change that.
hi, i am skilled voip developer. I have ready solution with webrtc2sip server based on kamailio + rtpengine.
Your task actually much bigger then just use webrtc2sip server.
Contact me for discuss details.
Hi, I have 18 years of experience in VOIP and asterisk. I offer my services for your project. Please have a look at my profile to see all my experience details. I assure you of a high quality professional service. I am one of top voip developers available with freelancers. We can chat out details.
Hello, Good Day.
I'm Shahidur with 10+ years of experience in diverse IT fields & I would love to help you with the setup. We certainly have some obvious things in common to discuss & to do so a proper conversation using chatbox would be sufficient enough I believe. If & Once, everything looks promising & perfect, why not start working right at that instance!
I'll be looking forward to learning more from you in the shortest period possible.
Thanks & Best Regards,
Md. Shahidur Rahman
SYSTEM ANDMIN EXPERT
GOOD IN Asterisk
HELLO DEAR CLIENT
I have gone through your project details and the work is DOABLE since its within my area of EXPERTISE.I will tackle are required guideline to deliver desired goal Please consider my bid. THANKS