What causes one-way or no audio during a call?

One-way or no audio during a call can be caused by various issues within the SIP message. To diagnose the problem, please review the Session Description Protocol (SDP) and make sure you’re meeting the following requirements:

1. The media IP address in the SIP Invite or 200 OK messages must be configured to be a public IP address when connecting to the Bandwidth network. 
2. The SDP should not contain a=sendonly, a=receiveonly, or a=inactive attributes.
3. The audio codecs should not be mismatched. This would need to be determined by setting up a media capture between the Bandwidth client and the Bandwidth Technical Assistance Center (TAC) team to verify the audio codecs being used. As the customer, you can also set up a media capture on the WAN side of your network or the public side of your firewall to verify the audio codecs being used in the media stream. You may also open a ticket if you need additional assistance. 
4. The audio stream must be sent from the media port and IP address that was negotiated within the SIP message between the Bandwidth client and the Bandwidth platform. This is a symmetric NAT and needs to be configured on your side.
5. Your PBX must not be set up to send audio only when it receives an audio stream. If this is the case, you need to make a configuration change within the PBX to make sure the audio is sent initially to start the audio path.

Note: This article doesn’t cover all possible issues, but only the most common ones. If all of the above requirements are met but you continue to experience one-way or no audio, please open a ticket with the Bandwidth TAC Team.