Steps to connect to a SIP conference call:
The following steps allowed to successfully connect to a conference set up through the renavisio service. It is assumed that among the details to connect to the conference, a SIP address (in the form sip:email@example.com) is provided. Connection was successful with both a linphone and an ippi account (the latter is recommended by the Jitsi Desktop software).
Install the Jitsi Desktop software. This should be done through the package management software of the given distribution (dpkg, yum, pacman, ...). For example, sudo apt install jitsi for Debian. If the Jitsi client is not available on the given repositories, an alternative such as Linphone may be considered. In the case of Debian, as of January 2017, Jitsi is only available on the Unstable repositories, where it seems to have broken dependencies. A possible workaround is discussed at this page.
After having installed a SIP client, add your SIP account. Place a test call with renavisio by dialing the address:
When asked for a password, dial:
If you are connected to the test conference, then you are ready to call the provided SIP address for the real conference.
If a call drops down immediately or if it tries to reach the server without succeeding for a long time, it may indicate a connection issue. Try to disable encryption through the client settings. Calls may fail behind VPN (place a test call without VPN) or if a proxy requires client software (e.g. Jitsi) customization. In this case more information should be provided by the local network administrator.
If the software does not connect to the SIP account, check the correctness of the account credentials and make sure to remove all SIP software except the one in use (it may block the relevant network ports).
Install the CSipSimple or Linphone programs from Google Play or F-Droid repositories and follow instructions to add your SIP account. Dial the conference SIP address to join the conference (see the address in the section above to place a test call). Dial the conference password if required. If it does not work, try to remove encryption. Calls my fail behind VPN or a proxy (requires customization).
Session Initiation Protocol (SIP) is a voice/video call protocol that can be used over the Internet. It requires the creation of an account with a SIP provider. Many are available. Some offer more options than others, for example encryption support or the possibility to charge credit to call regular phones (calls between SIP phones over the Internet are usually gratis). The steps described here were tested using linphone and ippi accounts (the latter is recommended by the Jitsi Desktop software).
Once a SIP account has been created, it can be used with any of the several SIP client programs. A conference call should provide a SIP address that can be called with the client program.
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