RFC says that annex B is the default in the absence of the annexb=no
format string. Consider the alias "G729a" as annex A.
Closes#1951
Change-Id: I9a8483ee5520bf4688601123f1ebec4f1f480642
(cherry picked from commit 3df26cf2d0)
(cherry picked from commit 754837ea9f)
Reverses the side effect of I225b43dff
Restores consistency and fixes tests and very slow systems
Change-Id: Ib4285e7aae95cd58eccd1d3c4f84e5786b4b0810
(cherry picked from commit ffacde5748)
(cherry picked from commit e5a1aec247)
Different encoding on certain platforms (armel)
Change-Id: I481d7aafcdba078bb317a6c6bab1f0b88b37a7b7
(cherry picked from commit 35d3110612)
(cherry picked from commit 387cb2a0a7)
Fix for "opus<>opus+DTMF" auto test.
Receive when port is already known.
Fixes:
Use of uninitialized value $port in numeric eq (==) at ../perl/NGCP/Rtpengine/AutoTest.pm line 239.
Change-Id: Ic0089bae769ccafeb7145e703a6e21029da85a97
(cherry picked from commit 2f664bc72e)
(cherry picked from commit 3c1e6ea893)
Protect selected_sfd with in_lock.
Protect RTCP sending with in_lock and out_lock as appropriate.
Has the odd side effect of RTCP reports expected in tests to be sent one
packet later than before.
Closes#1966
Probably fixes#1927
Change-Id: I225b43dff8e8fbb938d3be6aad50249997615d77
(cherry picked from commit ffc539c0d8)
(cherry picked from commit 2e714ebedd)
Using `l->next` as a test for whether only one element is left in the
list is not reliable. Use the actual list length instead.
Update one affected test.
Closes#1961
Change-Id: I5773715700220cd762e61090bac941ddd33afd9b
(cherry picked from commit 02897ccb46)
(cherry picked from commit b38f4d81ce)
In some cases, in particular when multiple input codecs are transcoding
to the same output codec (e.g. audio + DTMF -> audio), it's possible
that one of the input decoders produces frames in a format different
from the one expected by the encoder. Add an encoder-side resampler to
compensate for that.
Change-Id: I74d55edf47ac2fa65f950e68f4a5975f1ab947bb
(cherry picked from commit 461f663e58)
(cherry picked from commit 57bf3f1002)
With reuse-codecs, we still need to place the codec from the SDP into
our prefs list, even if it's already present, as the format options may
have changed. Update one affected test case.
Closes#1921
Change-Id: I688c57a8c45ec4c3bf159fe2193a0e00bbceeda2
(cherry picked from commit 65bfe22d40)
(cherry picked from commit 789971d929)
NGCP-Flow: mr12.5.1
Only meant for mr12.5+ branches.
Add a support of session level connection replacement by default,
avoiding flag.
This is due to the fact: we've managed to deprecate
the replace-session-connection functionality, but the switch
to the SDP create approach hadn't been done that time around,
when branching the master to the mr12.5 branch.
Change-Id: Ic5230180e3ba83cda59e62c567414c71b2d2f1aa
If media type doesn't match, assume a change in a=mid and ignore it.
Change-Id: Id0c8d7b9163445266e4179f4388109954226c4fa
(cherry picked from commit 2aa11c2a2a)
(cherry picked from commit 6480bd6284)
Fix zero c= line when first m= line has a zero port.
Add matching tests.
Change-Id: I1472b8101ffb3647e6aa092995672788510ee117
(cherry picked from commit 5b69e419d5)
(cherry picked from commit 28a2f02a0a)
The option flag `replace-origin-full` should replace whole
origin (o=) line of the SDP body, so that all origin fields
in the `o=` line always remain the same in all SDPs going to a
particular RTP endpoint.
So, not just a straight-forward replacing with own values.
A behavior in relation to the address field is the same
as by the `origin` option flag.
Correct according tests and add one more.
Change-Id: I4bcb740db5cd68a75fb754f5f6b5d21e19b05e1b
Use the `->session_last_sdp_orig` for the version as well as
username type of replacements.
Remove superfluous code.
It's possible to handle it like that, because the last user
of the sdp_replace func, which is the offer/answer model, sets
the `->session_sdp_orig` actually in `__call_monologue_init_from_flags`.
Additionally: add tests for the user type of the origin replacement.
Change-Id: Idd60e628c676840c2e844077303035c0edec67e2
Make the `sdp_replace()` compliant with `sdp_create()` in terms
of the replace `origin-full` flag support.
Also introduce according unit tests.
Change-Id: Ib950a957d5ee3820dbc8a831fe0a7ea570b4c59b
Instead of using separate data members to
bring data with the `call_monologue` structure,
just use the whole `sdp_origin` object type (structure)
as a pointer and keep it aling with real SDP origin
of according monologue's side.
Refactor the code accordingly for `sdp_create()`
users (firstly only here).
Additionally introduce functions to alloc/de-alloc
`sdp_orig` object:
- `sdp_orig_dup()` returns a pointer to copied object
- `sdp_orig_free()` deallocates it
Change-Id: Iff6a777e4867e78c73ca79c73fdb73ff8e9f22eb
Move the `call_subscribe_request_ng()` fully to a usage
of the `sdp_create()` only.
Carry the origin IP and net family via flags
to the monologue, so can be reused later when creating SDP.
Always use given SDP session origin IP address and family
for the SDP being prepared, unless sdp origin replacement
is required via given flags (in this case just used
an advertised IP of rtpengine).
Additionally: fix unit tests for subscribe cases accordingly
to the policy.
Change-Id: Ib7697876ce45e01597edd27764d4147d12f738c8
Run the SDES reordering routine also when the opposite side is plain
RTP. Add a matching test case.
Ref: https://groups.google.com/g/rtpengine/c/pG6NB24G-BY
Change-Id: I7844290e85e44c1bba2c7665202b6259aead1028
Split the code that handles reordering offerer SDES suites based on the
respective option into its own function. Rework it slightly so that it
reorders the list in place for simplicity. Remove the part that honours
SDES-no and SDES-only as these should affect the opposite (outgoing)
direction. This requires changing one of the test cases, which seems
more correct now.
Change-Id: Ie284d052d72031fad64c94767fa95c74639ae331
Don't carry parameters required for processing
in the `sdp_create()` via the `stream_params`,
but rahter handle them like:
- parsing in `sdp_parse()`
- `sdp_media` -> `stream_params` in sdp_streams()
- `stream_params` -> `call_media` in ` __media_init_from_flags()`
Additionally: update the test "subscribe_request AMR asymmetric".
This is because we seem to never actually take into account
presence of bandwidth data in offer/answer model preceding
the subscribe request.
Change-Id: I5b4b19ae244c6bbf961d5ea7c18b6747519144db
To prepare the deprecation of session level connection
usage (so the `c=` attribute of the session level),
deprecate first the `replace-session-connection` flag
support, because it will have essentially no use.
Change-Id: I9f9cb8f06d03d1c107a5c928054a79409642bb5b
Rework the `sdp_out_add_origin()` func so, that
it's compatible with offer/answer model.
Additionally it uses the previously existing logic
for the PUBLISH operations always, regardless any
conditions.
Additionally: fix auto-daemon-tests-websocket accordingly.
Change-Id: I0bddc67f4ebe47a24063ccc82428929aeb6cc37a
Always ensure to set the actual sdp session name when
creating SDP using `sdp_create()`.
A new dedicated func `sdp_out_add_session_name()` introduced
for that.
Additionally: fix auto-daemon-tests-websocket and
auto-daemon-tests-pubsub accordingly.
Change-Id: Ie62573149ef9ae226dc8c955dcb2cfaaa4d3ae87
now that the returned last_event_ts is always that of the previous
DTMF, we can ensure that the next one isn't transmitted until that
time plus the required pause.
Like the num_samples calculation, the actual time needs to be
increased by 1 packets worth of samples so tha pause lasts the
full duration required
Change-Id: I6da1dd7cbcf49f7f0431a5123df2cdc382fe3dba
this function is used to determine if a pause is needed on a new
injected DTMF's start ts to ensure a gap between the events. However,
if an inject request comes in after the end of the previous event
but before it would have been offset due to pause, no pause is added
This change returns the ts value from dtmf_state if the queue is
empty as that will always be the ts of the last DTMF transmitted
Change-Id: I4f3cf5115d1a8e26c0ca1bc9570c46e29391e0d0
the num_samples was added to the start_pts, which is the first event
packet timestamp, which has already increased its ts by its event
duration. so, the total duration of events ends up being one packet
more than intended.
Change-Id: I423bb222a81c5bd78e570ff2026c72dd4dd1b100
Make sure we increase the output RTP sequence
number for each generated packet in the case
of packets that need to be duplicated or sent
repeatedly (DTMF end event)
Change-Id: Ia16ffefc0791d01575248ac5d8025eb30ccaec67
Consider call_media SDP attributes as ones that were received instead of
the ones to be sent out. Use media subscriptions to look up the source
media and print that one's attributes when creating an outgoing SDP.
Change-Id: Ibdf3a77a6f8a61654e0fc7c14aae16dfc6eabf14
Fix up tests to match reordered attributes.
Remove 10-year-old a=ssrc parsing code completely as it's unused.
Change-Id: I78064b1b2f27a442fe8bf4b448c7174c5458d3b1
During an offer, we update the codecs from the given list not only on
the side of the offerer, but also on the answerer's side, in order to
perform the codec answer routine during the answer phase. While doing
this, we empty out the existing list of codecs (on both sides) and
repopulate it fresh from the given list.
This can cause problems during a reverse re-invite, when the list of
codecs on the answerer's side already contained the codecs that had been
offered before. When setting up the new re-invite offer, we want to
retain codecs (and their payload types and format parameters) that were
already in place, instead of recreating a new list from scratch.
Improve this by adding a `merge_cs` option to the populating functions,
which points back to the stream_params codec_store. Codecs that would
have been removed from the codec_store during the repopulation are then
moved back into the stream_params codec_store instead. This then allows
the functions adding new codecs to the list (offer/transcode) to
reference these codecs that were previously in place, and so they can be
added back with the same options as they had existed before, instead of
recreating them from scratch.
Change-Id: I53e7ab10e9144a308a5c36be5ebfddd73c212f06
Move `a=extmap:` attribute to the `call_media`'s
`sdp_attributes` using `stream_params`.
Required by the `insert_sdp_attributes()` used in `sdp_create()`.
Additionally revert `extmap` unit test checks in the
`testVideoroomWebRTCVideo`.
Change-Id: If63e4e8733ea0899f34fae1f1d38997e9e2c081c
Richard Fuchs
Donat Zenichev
Tom Briden