New rtp.io module (embedded RTPProxy) #3501
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Need to work around issue with pushing to ghcr, so pipeline works in the PRs. |
What rtp.io module does is it starts up a RTP handling threads in the main process and let rtpproxy module access those threads via a 1:1 socketpair, thus providing usual media relaying functionality without using any external relay process. The module requires RTPProxy 3.1 or higher, compiled with --enable-librtpproxy. When rtpproxy module is loaded without arguments and rtp.io is loaded as well, the sockets exported by the rtp.io will be used automatically in the set 0. Otherwise those sockets can be used as part of other set(s) by using "rtp.io:auto" moniker. Provided is also a CI job that builds rtp.io in a rtpproxy container for all supported architectures and tests it using voiptests package. TODO list: o hook up notification socket directly into appropriate handler on the OpenSIPS side; o generate -l / -6 option(s) based on OpenSIPS's own socket list; o more documentation.
Use rtp.io when no explicit rtpproxy configuration is provided. It is also possible to mix internal RTP functionality with externals proxies by adding it up as "rtp.io:auto" into some of the existing set.
This fixes build on systems that have no backtrace API in the libc (e.g. BSDs).
CI: handle debian.
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Summary
This PR includes a new rtp.io module which allows to provide media relaying and other RTP manipulations without external media relay.
Details
Traditionally, OpenSIPS only handles signalling, offloading media operation to the external relay. However in some situations using external relay may be undesirable from system complexity point of view and others, so having an option to handle it internally may be useful.
Solution
What rtp.io module does is it starts up a RTP handling threads in the main OpenSIPS process and let rtpproxy module access those threads via a 1:1 socketpair.
The module requires RTPProxy 3.1 or higher, compiled with --enable-librtpproxy.
When rtpproxy module is loaded without arguments and rtp.io is loaded as well, the sockets exported by the rtp.io will be used automatically in the set 0. Otherwise those sockets can be used as part of other set(s) by using "rtp.io:auto" moniker.
Provided is also a CI job that builds rtp.io in a rtpproxy container for all supported architectures and tests it using voiptests package.
Compatibility
None
Closing issues
None