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FAQ.md

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How to report issue?

We receive tens of messages on our dev-groups and private mailboxes every day. Without a commercial license, we only provide a best-effort support on doubango-discuss.
We're happy to help you to fix your issues but we'll not spend hours on them to understand what's wrong. If you want help, you must:

  • Provide clear technical description of the issue.
  • Change you config.xml to use INFO debug level.
  • Attach the logs to the message (do not cut the logs).
  • Provide information about the SVN revisions (both Doubango and webrtc2sip). If the report is about building issues:
  • Attach config.log files for both Doubango and webrtc2sip you should also:
  • Provide the network (Wireshark) capture at the server-side
  • Provide the browser logs.

How to create self-signed SSL certificates?

We'd recommend reading this thread.

How to redirect the output to a file?

webrtc2sip logs the messages to sdterr and stdout. To redirect the logs to webrtc2sip.log:

webrtc2sip >webrtc2sip.log 2>&1

I see "Remote party requesting DTLS-DTLS (UDP/TLS/RTP/SAVPF) but this option is not enabled". How can I fix this

DTLS-SRTP is required by some WebRTC implementations (e.g. Firefox). You MUST:

  1. use a new OpenSSL version with support for DTLS-SRTP as explained here. Linux almost always comes with OpenSSL pre-installed which means building and installing OpenSSL by yourself will most likely duplicate it.
  2. make sure you don't have more than one OpenSSL version installed (look for libssl).
  3. rebuild webrtc2sip and make sure the "CONGRATULATIONS" message says that you have DTLS-SRTP enabled.
  4. make sure you're using SSL certificates in your config.xml (see technical documentation ). DTLS-SRTP requires at least a valid Public Key (could be self signed).

I see "Too many open files" error when I try to connect a client. How can I fix this?

If you're using webrtc2sip 2.5.1 or previous:
On CentOS, Fedora, Redhat and many other Linux ditros the number of sockets an application could open is limited to 1024. To check your limit:

ulimit -n

For example, to change this limit (You must be carefully) to 2048:

ulimit -n 2048
# then restart webrtc2sip

If you're using webrtc2sip 2.6.0 or later:
Change max-fds entry in config.xml.

I see "a=crypto in RTP/AVP, refer to RFC 3711" on my console when calling FreeSWITCH. How can I fix this?

Edit the config.xml file and change the srtp-mode from optional to mandatory.
Starting version 2.1.0, it's no longer required to change config.xml.

Why do I have delay on H.264 stream from webrtc2sip?

When the media coder module is enabled we use FFmpeg and libx264 to encode the stream. libx264 uses presets to define some default configuration parameters. Setting these presets from FFmpeg wrapper seems to be useless as the values are never forwarded to libx264. To fix this issue, you need to apply the patch at http://code.google.com/p/doubango/source/browse/branches/2.0/doubango/thirdparties/patches/ffmpeg_libx264_git.patch
Starting Doubango 873, it's no longer required to patch FFmpeg