SIP Protocol / Handshake Backend FlexiSip Server Debugging Required

Job ID: 33203763

Budget: $15 – $25 USD

We have a WhatsApp-like app that we developed. It is much more secure than other chat apps. We have added secure audio and video to our app based on the open-sourced Linphone frontend and FlexiSip server backend.

The dev team has integrated Linphone and FlexiSip server; however, we are unable to get the audio and video to work properly (i.e., we cannot hear the audio nor see the video when making calls). It does work from iPhone to iPhone offshore where the dev team is located, but not in the U.S. Linphone and usually uses a cellular mobile number as the basis for making calls and associates the SIP ID to the phone number. We do not use mobile numbers for privacy reasons so we create the Sip ID during registration; however, this properly (i.e., the user is assigned a SIP ID when they register, and that is used for making audio and video calls).

Our dev team members are not SIP experts. An expert who knows SIP protocol / handshake, and who knows the FlexiSip server should be able to help our dev team figure out the audio/video problem (i.e., why the audio cannot be heard and video cannot be seen for certain geo locations).

This will start as a fixed bid but can be made an hourly project as well.
Related categories: Linux Mobile App Development iPhone Android SIP