mirror of https://github.com/sipwise/sems.git
master
mr26.1
mr12.5.1
mr12.5
mr14.0.1
mr13.5.1
mr26.0.1
mr14.1.1
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pristine-tar
upstream
mr3.8.2
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mr3.3.1
1.4.3+spce2.8
mr3.2
mr3.2.2
agranig/notify
mr3.2.1
1.4.3+spce3.0
vseva/dump_timers
sayer/1.4-spce2.6
upstream/1.6.0
upstream/1.6.0_20150612_a2d214df36
upstream/1.6.0_20150519_ae8a42
upstream/1.6.0_20140819_0b6d12
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ngcp/1%1.4.3-1
${ noResults }
8 Commits (master)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
8e1a41a9ec |
MT#59962 core: introduce new RTP transport model
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
|
5 months ago |
|
|
8083dcd92f |
MT#62181 use shared_ptr for msg_logger
Use shared_ptr for reference counting instead of hand-rolling it. Change-Id: I147dd6df4e302a2400b184aa88775cf635a57506 |
10 months ago |
|
|
ede52ab0b3 |
MT#61912 OA: save the established body when it's generated
When OA generates own SDP body (e.g. case with accepted invite in DSP), save it as established_body for this leg, in order to be able to re-invite this leg later. It can happen there is still no SDP seen from the callee side (other side), and in case we want to send established re-invite towards this leg, this will fail, since no SDP body seen yet. Change-Id: Ifd9f0fb70d27deac871de4eed1648f7c152813f3 |
1 year ago |
|
|
cdcbbb7866 |
MT#57562 sems (CE): Add Allow header in 200 OK when empty
This is a partial rework of b67428d. The patch was always saving the Allow header from the initial INVITE and replacing the Allow header in the 200 OK with that. This fixed an issue with AA, where Allow header in the 200 OK, generated by the AA, was empty. But it causes an issue in case of normal calls between A and B. In general we want to keep the Allow header in the 200 OK, if exist. Instead, we want to add the Allow header in the 200 OK response in case the 200 OK does not contain the Allow header. In this case we won't set the original Allow from the INVITE but instead we will set a default Allow header (SIP_REPLY_DEFAULT_ALLOW) containing all the major SIP method. Change-Id: I6bf5a2cc7e0228f7b99d28963d16c14c463be695 |
3 years ago |
|
|
aec816dad7 |
MT#56321 Add 'Allow:' header to the 200OK, if present in the request
We have to take care of keeping the actual 'Allow:' header in the request, if the incoming request has had it. One of the use cases for that is for example the INVITE coming to the AA, when we must respond with the 200OK, but there is no 'Allow:' header inside the 200OK. Certain systems can be confused by that, even though the RFCs are not forcing us to do so. Which can even lead to an impossibility to use a normal call clearing with the BYE method, from behalf of the caller. Change-Id: I9bafb89e4aa5e53077450068f847c07a497b43f1 |
3 years ago |
|
|
2bf464e5cb |
Imported Upstream version 1.6.0~20150612~a2d214df36
|
11 years ago |
|
|
be6b7d527c |
Imported Upstream version 1.6.0~20140819~0b6d12
|
12 years ago |
|
|
df9992620d |
Imported Upstream version 1.6.0~20140516~eaa616
|
12 years ago |