One Way Audio
One Way Audio
The one way audio is usually an issue experienced between two or more SIP/IP desktop or Web/Softphones. When one or more SIP phones on the established calls are not able to hear each other, or one of them is unable to hear the other party, this is a typical case of inaudible phone call on either sides. There can be some basic issues which one can look in order to avoid such issues. Please also refer to this doc page for information on ports that need to be open on Vodia PBX.
Handful number of issues that can be avoided
Go to your local router (or the router which is hosting the Vodia PBX) and make sure the RTP ports are unblocked or are listening on the ports 5060 (by default or any another predefined port). You can verify the open ports using the commands like:
Linux systems:
netstat -atn (for TCP) netstat -aun (for UDP) netstat -atun (for both)
For Windows System:
netstat -aon | more
Lets say that the IP phone facing the issue is "Yealink" system (Vodia PBX supports the provisioning of Yealink SIP-T28P/SIP-T26P/SIP-T22P/SIP-T20P/SIP-T46G series phones). So, type it's web address (IP address or the alias name) on the web browser and open up the login page for "Yealink" and login with your default login. Under Network go to Advanced and choose the same range of the RTP ports which were provided under the Audio option inside the Settings in Domains on Vodia PBX as below.
Port range start - 49152
Port range end - 64512
Advertising the correct public IP address
To ensure two way audio, make sure to advertise the correct public IP address. If your system is on a NATed network (or behind a router), advertising a private IP address in the SIP signaling to fellow servers may cause one way audio. Most systems will allow configuration to advertise the correct public IP, either by statically configuring the address, or by the use of a STUN server to discover the current public IP address. You can find that settings in Admin > SIP > Settings as shown below.
SIP ALG
SIP ALG (Application Layer Gateway) acts as a media between your system and the Flowroute. It's main purpose is to assist the Vodia PBX and the IP phones behind a NAT service. Hence, they actively monitor and often modify the SIP packets. Poor implementation of SIP ALG often cause one-way audio, dropped calls, run-away calls etc. Make sure each of your firewalls and/or routers for any SIP ALG settings and disable it.
Unsupported Codecs
Vodia PBX supports six types of RTP codecs namely, G.711U, G.711A, G.722, G.729A, G.726, GSM 6.10 currently. If your systems doesn't support any of the above codecs, then you might end up hearing one-way audio between systems / IP phones.
Points to be noted for a successful WebRTC call
Make sure that you are running a latest (66.0 or higher) version of the Vodia PBX on your system.
The call has to be made by switching to the user mode of the extension on the Vodia PBX. Importantly, the URL on which your PBX is hosted should have "HTTPS" instead of "HTTP" in it. As, the valid certificates are needed on the server in order to have a WebRTC call setup successfully.
The Firewall running on the system hosting the Vodia PBX, should not block any of the ports used for the WebRTC call. Various firewalls from different vendors will have varied settings which have to be taken care of so that it won't block the SIP traffic.
Vodia Support
These might be some of the points you can check off the list before worrying about the one way audio call to be a bigger issue than expected. If the above points doesn't prove of a big help, Vodia Networks would be glad to extend support for it. You can open a ticket with Vodia Networks by clicking on this URL https://vodia.zammad.com/#login or you could also check the Vodia Forum for more details of the problem (if existed in the past) http://forum.vodia.com/.