Asterisk & webrtc based project. - open to bidding

Lets discuss to check this issue and fix it. One way audio are result of NAT, etc. So, lets check the setup and fix.

Hello, I can do the job. If you want we can discuss the project details via PM. I also speak spanish. Be free to contact me if you have any questions or comments about it. Thanks in advance. -- Adolfo

A proposal has not yet been provided

I have a great deal of experience with Asterisk, Freeswitch nad Webrtc. Friend of mine acutally told me about this bid and that made me join freelancer. I would like to know if you are interested. I have no prior experience here in Freelance I should mention. By the way, Webrtc being an experimental product and not mature enough.I would recommend oversip as alternative. Thanks.

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.

Hello, I've some years (+11 years) of experience on the Linux administration field, I'm new on Freelancer and maybe my profile isn't very complete... But returning to your issue, some times the voice in one way (instead of two) is caused by firewall or routing problems. Asterisk uses SIP with UDP (port 5060) transport for the session initiation and so on for the voice transport it uses RTP fom port 10000 to 20000 (in most cases). When a problem as yours occurs, the first reason can be because the above described port configuration isn't comunicating fine. Other possibility is the use of incorrect codec or voip client configuration in one of the ends. Analyzing the environment is possible find the gap and solve it. Have a nice day. Thank you, regards.

This can be as simple as firewall issue of can be internal context, we can help u with this issue and we guarantee our work full money back guarantee if not happy.

A proposal has not yet been provided