This commit introduces a new RTP transport model,
based on the AmRtpTransport abstraction, providing a more modular and
efficient approach to RTP handling compared to the legacy model.
General things:
- Introduction of new RTP receiver and transport abstractions
- The new model coexists with the legacy RTP handling
to preserve backward compatibility
Architecture overview:
- A new AmRtpTransport layer is introduced to encapsulate RTP transport logic
and decouple it from AmRtpStream and session-level code
- Transport-specific functionality is implemented via dedicated classes:
- AmRtpTransport: base abstraction responsible for RTP/RTCP packet flow,
lifecycle management, and interaction with streams
- AmRtpUdpSocket: UDP-based RTP socket implementation
- AmRtpSocketPair: manages RTP/RTCP socket pairs and their coordination
- AmRtpStream is refactored to rely on the transport interface
instead of directly managing sockets and low-level I/O
Integration details:
- SIP session and dialog logic (AmSession, AmSipDialog) is updated to work with
the new transport model
- SBC call legs and call profiles are extended to support transport selection
and lifecycle handling
- SDP processing (AmSdp) is updated to reflect transport capabilities and
address handling changes
This refactoring improves separation of concerns, reduces coupling between RTP
streaming and transport mechanics, and provides a cleaner foundation for future
RTP extensions and optimizations.
Code provenance and license:
- part of the code was written by Fokus GmbH
- part of the code was written by Sipwise GmbH
- part of the code base copyrighted by Fokus GmbH was provided by them in
the form of backports under the GPL license.
- these changes are covered by the GPL, matching the existing project license
Change-Id: I313c4abd9b54f805c1e668c019cf6fb8c4a9e46d
Add missing initialisers and fix order.
In some cases the member was actually unused and could just be removed.
Change-Id: I0f0c927eb8271c35dcfd371f225847f62bea2812
Warned-by: Coverity
Backport from the upstream to add `lock(mutex)` calls
in order to avoid crash.
Upstream commit: 0af0c924a8fc9638c5b18a4d5799a085ea8132bc
Change-Id: I022b09d2827b41f0785a9c0568c334b8a2af108f
We have to add support for playing an early announce for those
failed trasnfers, which for example reach the timeout.
This means there is a need to do the following:
- send 183 from early announce DSM module
- receive 183 on the B2B leg, which was sending an INVITE to
the transfer destination
- update the caller with an actual media capabilities,
so it is able to receive the early announcement
The mechanism behind that is that:
- if Proxy in the other leg, which was used to reach the party
(to which the transfer has been done),
requests the force usage of the announcement for the caller
in the very first leg (with the caller who is on hold now)
- then the DSM functionality adds the 'P-Early-Announce: force' header in 183
to let sems-b2b in the very first leg know, that there is a need to embed
the audio from the early media into the media session with the caller.
- hence, caller gets a media re-negotiation
- and the leg with the DSM doesn't get updated, since it's still remains
in the Early stage of the dialog
All the solution is built around two important things:
- Proxy uses the P-App-Param called ';force_early_announce=1' to show to the
DSM that it needs a "forcive" early media, which must "override" the MoH
for the caller, if it's being played.
- Sems-b2b uses the header 'P-Early-Announce' to let the very original leg
with sems-b2b know, that there is a need to embed early media into
the already established media session with a caller
Original ticket number: TT#187351
Change-Id: I2cda231e877d9ba91eaa1738322b0981618c1dbf
Because of messy organizing of the code in the function,
it's not possible to read that and work with that.
Original ticket number: TT#187351
Change-Id: I3c92889d44c03b11380fc923e0b22c1ec8961380
Donat Zenichev
Richard Fuchs
Victor Seva