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New rtp.io module (embedded RTPProxy) #3501

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@sobomax sobomax commented Oct 18, 2024

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

@sobomax sobomax force-pushed the wip_rtp.io branch 3 times, most recently from 0464fea to 26f9391 Compare October 21, 2024 17:20
@sobomax sobomax marked this pull request as draft October 21, 2024 17:23
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sobomax commented Oct 21, 2024

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.
@sobomax sobomax marked this pull request as ready for review October 23, 2024 21:47
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Any updates here? No progress has been made in the last 30 days, marking as stale.

@github-actions github-actions bot added the stale label Nov 23, 2024
@github-actions github-actions bot closed this Nov 30, 2024
@vladpaiu vladpaiu reopened this Nov 30, 2024
@stale stale bot removed the stale label Nov 30, 2024
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2 participants