Asterisk & webrtc based project. -- one way audio in many incoming calls

My bid is for figure out what is going not well and fix it, if problem doesn't need changing of the whole system. Our team is specialized in VoIP development, VoIP and Asterisk especially is our main specialization. We have wide experience with WebRTC, webrtc2sip and SIPML5. We are located in UTC+2 timezone.

Most of these one-way audio is because of non-proper NAT setup. 3 days and I'll solve it for you. Payment only done after solution is given, very simple. Check my reviews, and ask any questions here or Skype (nttranbao) if you need before we start. I can start within today, so dont hesitate. Kind regards, Bao Nguyen

1. NAT Traversal 2. Firewall 3. Codecs What Asterisk version are you using? Let me know and we can go through. Regards, Tomas

Hello, a few words about me: I manage over 300 servers with different purposes: web servers, sql servers, voip servers, filesharing servers, email servers and so on. (most of server are used as voip servers with different purposes) Each server has its own configuration, monitorization, backup etc. If you are interested I can give you more details about my work. about your project: I do this kind of things for a few of servers that I manage. If I will be your choice I will offer you my support for other projects. Can we talk a little bit please? I need more informations. Thank you. Depending on the problem my bid can be lower than it is right now!

Hi there, I've been working with Asterisk and VoIP overall for years. Ready to help you with solving one-way audio issues. Thank you.

Buenas tardes, Soy técnico certificado en Asterisk (DCAP) by Digium, con lo que puedo ayudar a resolver el problema de falta de audio. Muchas gracias, Saludos Iván

Hi, I have more than 9 years of experience of linux admin with lots of flavors of posix operating system like Redhat linux, centos, debian ubuntu, suse etc and shell script too.I have about 4 years of experience of VOIP. I can solve this issue, many times this issue comes from DTMF error. I need remote access through ssh and testing setup of your soft phone. If you are using x-lite than need a test extension.

I have a lot of skills with voip service like, SIP Servers, IP-PBX, UA and SoftSwitches. Please look my profile and you can see it.