Opensips as Webrtc Gateway for Asterisk
$15-25 USD / hora
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 Opensips 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.
Nº del proyecto: #28121116
Sobre el proyecto
4 freelancers están ofertando un promedio de $19 / hora por este trabajo
Hi, there! I can provide WEbrtc2SIp gateway for you because i did that many times. Contact me for more details. Also, you need some custom staff to control attackers and so on.
Hi, I am an engineer with more than 15 years of experience with VoIP projects (Asterisk, FreePbx, Issabel, Opensips, SMS, Call center, Wss) I can give you support in the configuration of your project, let me know more Más
Hi we are a team of experts with more than 12 years of experience in telecommunication apps, PBX systems especially in asterisk , sip and nodejs. We also have experience in distributed telephony systems using Kamailio Más
