When initializing a media for the other side (while
processing an offer), keep not only the `->protocol`
updated, but also the `->protocol_str`, so that this
can be usable while preparing an SDP for it.
Change-Id: I209b9048fae836903c9165c6b87682a06e77a744
Remove all codecs that cannot be transcoded to when the audio player is in
use.
Add safety to make_transcoder to return failure in case an unsupported
codec is requested for transcoding.
Convert leftover passthrough handlers to SSRC passthrough if there are
any (shouldn't be).
closes#1858
Change-Id: I1822e48723622d550624c7355a1acfbf8ca38eb8
Just keep the `s=` attribute empty, if it is
coming like that in the message being processed.
Current behavior of the sdp_create approach just
inserts the `rtpe_config.software_id`, which is wrong.
Change-Id: I1235081b90786f5d9480fd1cad9f2751c60e7720
Instead of setting it in two lines via temporary variable,
just negate it in one line for laconicity reasons.
Change-Id: Ie65d68ef5b8082f5de544ab2a9833a59d5e01d9a
Added support of the CT (conference total) bandwidth
attribute for the SDP session level. See RFC8866.
Will be required later for the SDP formatting in `sdp_create()`.
Change-Id: Ifc64f68d7acee8ce253882f4fa480bbf7ad7c0bd
Even if the media isn't accepted, still check for the mid
presence and add it, if required. This is the only attribute
to printed, for the media which wasn't previously accepted
(so 0 port added for it).
See: #1361 and #1362.
Change-Id: Ib92930d6e1ec4dfe526fa18869c5f7a8e17f6a4f
Move handling of the usual `m=` line, so printing
of: media type, port, protocol (e.g. RTP/SAVP),
to a separate function to make the `sdp_create()`
implementation be more laconic for reading.
This function is a default alternative for the
`sdp_out_add_osrtp_media()`.
No functional changes.
Change-Id: I469abbbf6e203d2cc655a26bbf44ff3c7a66b1df
Move a check that port isn't 0 to the function itself,
instead of checking each time before calling the func.
Change-Id: Iec43770fc0f5846c63f6e11d8c2dfb2be135aec6
Make the `sdp_create()` implementation be more laconic,
and move the OSRTP related handling into separate funcs:
- `sdp_out_handle_osrtp1()`
- `sdp_out_handle_osrtp2()`
Just refactoring, no functional changes.
Change-Id: Ib3c5f58d215ea3e0622d328656d82919dfa391c5
Move printing of other session level attributes (so taken
from the coming SDP and to be carried out to the outgoing one)
to a dedicated function `sdp_out_add_other()`.
No functional change.
Change-Id: I94713545e7610d74d36a6a5f3027c64407c6750c
Add a function to handle other non specific attributes
for the session level.
E.g.: ice-lite or anti-looper (rtpengine).
Change-Id: I147403ff9e3c3ff8be6718b8b118e405442a9220
In case when the `packet_stream`, selected via the
media being iterated, has no `->selected_sfd`,
just set the port value to 0.
Also skip adding any further attributes, like
connection information or any other, since
this media by fact is not accepted.
This is a usecase for OSRTP related scenarios,
where RTP media session can have 0 port set.
Change-Id: I7784b80e7d51d19106ea30db585143375a5ba050
... so that the desired wav channel can be controlled when producing a
mixed audio file
When a mixed wav file is created, the channels in the wav container are
currently allocated in the same order as each SSRC is received, meaning
it is impossible to know which channels have been allocated to the offer
or answer side of the call. Furthermore if there is a reinvite or media
file played, these are also allocated in the order that SSRC is received
- so an "answer" could end up sharing a channel with an "offer" with no
way of knowing this.
This patch allows you to specify how many channel slots should be
allocated within the mixer, and allows you to then specify which slot is
assigned to each media in the call (this will usually be 2 slots in
total, slot 1 for answer, slot 2 for offer or vice versa).
Ported from https://github.com/sipwise/rtpengine/pull/1852Closes#1857Closes#1852
Change-Id: I010208427cabc3a48d6ef7bd3a84e9a5bdcfd492
If a stream has been pushed to the kernel from anything other than RTP,
even though RTP is expected, we get a forwarding entries without any
SSRCs. This is valid, but once actual RTP is received, it needs to be
passed on to user space, so that SSRC contexts can be set up.
Possible fix for #1855
Change-Id: I51b82d3cf79cf66780fdde154bebe56e0f43174b
Move printing of attributes to a separate func,
in order to be able to re-use this print during
the same media section iteration (e.g. with OSRTP).
Later on this func is to be merged with the
`print_sdp_media_section()` function, after
the sdp_replace is fully deprecated.
Change-Id: I15825ed6eecb450b742b31677961fd8ddbf950fb
If there are no more tokens, instead of returning a null string as
remainder, set the remainder to the end of the original string with a
zero length.
Update two places in the code where this makes a difference.
Change-Id: I137fbce3e7d88ccba65f6b23a36aa1dfbbd3867e
The subscriptions must be removed first and the update function must be
called second. Otherwise unsubscribe doesn't have any immediate effect.
Closes#1851
Change-Id: Iccc449357f7136b863ba2d633b24942dbe87d767
Donat Zenichev
Richard Fuchs
Peter Kelly
Morten Tryfoss