Some places in Asterisk did not treat the formats on a stream
as immutable when they are.
The ast_stream_get_formats function is now const to enforce this
and parts of Asterisk have been updated to take this into account.
Some violations of this were also fixed along the way.
An additional minor tweak is that streams are now allocated with
an empty format capabilities structure removing the need in various
places to check that one is present on the stream.
ASTERISK-28846
Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe
This change fixes a few re-negotiation issues
uncovered with fax.
1. The fax support uses its own mechanism for
re-negotiation by conveying T.38 information in
its own frames. The new support for re-negotiating
when adding/removing/changing streams was also
being triggered for this causing multiple re-INVITEs.
The new support will no longer trigger when
transitioning between fax.
2. In off-nominal re-negotiation cases it was
possible for some state information to be left
over and used by the next re-negotiation. This
is now cleared.
ASTERISK-28811
ASTERISK-28839
Change-Id: I8ed5924b53be9fe06a385c58817e5584b0f25cc2
Based on this new endpoint setting, a joint list of preferred codecs
between those received from the Asterisk core (remote), and those
specified in the endpoint's "allow" parameter (local) is created and
is used to create the outgoing SDP offer.
* Add outgoing_call_offer_pref to pjsip_configuration (endpoint)
* Add "call_direction" to res_pjsip_session.
* Update pjsip_session_caps.c to make the functions more generic
so they could be used for both incoming and outgoing.
* Update ast_sip_session_create_outgoing to create the
pending_media_state->topology with the results of
ast_sip_session_create_joint_call_stream().
* The endpoint "preferred_codec_only" option now automatically sets
AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref.
* A helper function ast_stream_get_format_count() was added to
streams to return the current count of formats.
ASTERISK-28777
Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437
RFC5621 requires any content type with a Content-Disposition
with handling=required to be rejected with a 415 response
ASTERISK-28782 #close
Change-Id: Iad969df75936730254b95c1a8bc3b48497070bb4
When an outgoing channel is created a list of formats may
optionally be provided which is used as a request that the
formats be used if possible. If an endpoint is not configured
for any of the formats we ignore this request and use what is
configured. This has the side effect of also including other
stream types (such as video) that were not present in the
requested formats.
This change makes it so that the intention of the request is
preserved - that is if only an audio format is requested then
even if there is no joint audio format between the request and
the configuration we will still only place an audio stream in
the outgoing call.
ASTERISK-28787
Change-Id: Ia54c0c63e94aca176169b9bae4bb8a8380ea245f
When the Asterisk receives 200 OK with invalid SDP,
the Asterisk/PJPROJECT terminating the session.
But if the channel was in the Bridge, Asterisk tries send
the Re-Invite before terminating the session.
And when the Asterisk sending the Re-Invite, it doesn't check
the SDP is NULL or not. This crashes the Asterisk.
Fixed it to close the session correctly if the UAS sends the
200 OK with wrong SDP.
ASTERISK-28743
Change-Id: Ifa864e0e125b1a7ed2f3abd4164187e1dddc56da
The state of the default audio stream is used for hold/unhold so we
restrict it to sendrecv as the core does not handle when it changes as
a result of hold/unhold.
This restriction does not apply to other media types though so we now
only restrict it to audio. This allows the other default streams to
store their state at all values, and not just sendrecv and removed.
ASTERISK-28783
Change-Id: I139740f38cea7f7d92a876ec2631ef50681f6625
Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that
specifies the preferred order of codecs after receiving an offer.
This patch does the following:
Adds a new enumeration, ast_sip_call_codec_pref, used by the the new
configuration option that's added to the endpoint media structure.
Adds a new ast_sip_session_caps structure that's set for each session media
object.
Creates a new file, res_pjsip_session_caps that "implements" the new
structure and option, and is compiled into the res_pjsip_session library.
ASTERISK-28756 #close
Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f
This change adds support to bridge_softmix to allow the addition
and removal of additional video source streams. When such a change
occurs each participant is renegotiated as needed to reflect the
update. If another video source is added then each participant
gets another source. If a video source is removed then it is
removed from each participant. This functionality allows you to
have both your webcam and screenshare providing video if you
desire, or even more streams. Mapping has been changed to use
the topology index on the source channel as a unique identifier
for outgoing participant streams, this will never change and
provides an easy way to establish the mapping.
The bridge_simple and bridge_native_rtp modules have also been
updated to renegotiate when the stream topology of a party changes
allowing the same behavior to occur as added to bridge_softmix.
If a screen share is added then the opposite party is renegotiated.
If that screen share is removed then the opposite party is
renegotiated again.
Some additional fixes are also included in here. Stream state is
now conveyed in SDP so sendonly/recvonly/inactive streams can
be requested. Removed streams now also remove previous state
from themselves so consumers don't get confused.
ASTERISK-28733
Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5
Given a scenario where session refreshes occur close to
each other while another is finishing it was possible for
the session refreshes to occur out of order. It was
also possible for session refreshes to be delayed for
quite some time if a session refresh did not result in
a topology change.
For the out of order session refreshes the first session
refresh would be queued due to a transaction in progress.
This transaction would then finish. When finished a
separate task to process the delayed requests queue
would be queued for handling. A second refresh would
be requested internally before this delayed request
queued task was processed. As no transaction was in
progress this session refresh would be immediately
handled before the queued session refresh.
The code will now check if any delayed requests exist
before allowing a session refresh to immediately occur.
If any exist then the session refresh is queued.
For the delayed session refreshes if a session refresh
did not result in a topology change the attempt would
be immediately stopped and no other delayed requests would
be processed.
The code will now go through the entire delayed requests
queue until a delayed request results in a request
actually being sent.
ASTERISK-28730
Change-Id: Ied640280133871f77d3f332be62265e754605088
A previous review, 13174, made a change whereby on an incoming offer SDP
the pending topology was initialized to the configured. This caused a problem
for bundle with WebRTC where bundle could reference a stream that did not
actually exist if the configuration had both audio and video but the
offer SDP only contained audio.
This change undoes that review and instead fixes the original problem it
sought to solve by setting the state of created streams based on the
contents of the offer SDP. This way the stream state is not inactive
until negotiation later completes.
ASTERISK-28659
Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355
We need to copy the endpoint name before we call ao2_cleanup() on it,
otherwise we might try to access memory that has been reclaimed.
ASTERISK-28445 #close
Reported by: Bernhard Schmidt
Change-Id: I404b952608aa606e0babd3c4108346721fb726b3
Found during some testing, there is a race condition between selecting an
appropriate bridge type for a call versus the applying of media on the callee's
session. In some instances a native bridge type would have been chosen, but
due to the callee's media not yet being established at bridge compatibility
check time the simple bridge type is picked instead.
When using chan_pjsip this initiates a topology change event. The topologies
are then compared for the two sessions. However, when the topology was created
for the caller its streams are initialized to "inactive". This topology is then
used as a base when creating the callee's topology, and streams. Soon after
the caller's topology's stream(s) get updated based on the sdp (get set to
sendrecv in the failing scenario).
Now when the topology change event is raised, and the two topologies are
compared, the comparison fails due to a stream state mismatch (sendrecv vs
inactive). And since they differ a reinvite is sent out (to the caller in
this case).
This patch makes it such that when the caller's topology is initially created
it gets created based on its configured endpoint's media topology. When the
endpoint's topology is created its stream's state(s) are initialized to
sendrecv instead of inactive. Subsequently, now when the callee's topology is
created its topology streams are now initialized to sendrecv. Thus when the
topology change event occurs due to the mentioned scenario the stream states
match for the given sessions, and the reinvite is not sent unless due to some
other valid mismatch.
Note, this patch only changes one pending media state's creation point. It's
possible other places *could* be changed, however for now it was deemed best
to only alter what's here.
Change-Id: I6ba3a6a75f64824a1b963044c37acbe951c389c7
Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows
the on-hold behavior to be controlled on a per-call basis
ASTERISK-28542 #close
Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8
In chan_sip, there was variable SIPFROMDOMAIN that allows to set
From header URI domain per channel. This patch introduces res_pjsip
variable SIPFROMDOMAIN for backward compatibility with chan_sip.
ASTERISK-28489
Change-Id: I715133e43172ce2a1e82093538dc39f9e99e5f2e
Currently, the Asterisk's pjsip_session module does not keeping the
rtcp's stats info after it was removed. But by adding the results
vector and keeping it until session is destroying, it can give more
useful information for other modules.
ASTERISK-28253
Change-Id: Ib25c2d3fc4da084aecfde2a82c1b1d733bd64fa5
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list. Remove ao2_container_alloc macro.
Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088
Add a new global flag to res_pjsip to allow the callerid to be used
as the username in the contact header. This allows chan_pjsip to have
the same behavour as chan_sip
ASTERISK-28087 #close
Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95
This change implements a few different generic things which were brought
on by Google Voice SIP.
1. The concept of flow transports have been introduced. These are
configurable transports in pjsip.conf which can be used to reference a
flow of signaling to a target. These have runtime configuration that can
be changed by the signaling itself (such as Service-Routes and
P-Preferred-Identity). When used these guarantee an individual connection
(in the case of TCP or TLS) even if multiple flow transports exist to the
same target.
2. Service-Routes (RFC 3608) support has been added to the outbound
registration module which when received will be stored on the flow
transport and used for requests referencing it.
3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been
added to the outbound registration module. If a P-Associated-URI header
is received it will be used on requests as the P-Preferred-Identity.
4. Configurable outbound extension support has been added to the outbound
registration module. When set the extension will be placed in the
Supported header.
5. Header parameters can now be configured on an outbound registration
which will be placed in the Contact header.
6. Google specific OAuth / Bearer token authentication
(draft-ietf-sipcore-sip-authn-02) has been added to the outbound
registration module.
All functionality changes are controlled by pjsip.conf configuration
options and do not affect non-configured pjsip endpoints otherwise.
ASTERISK-27971 #close
Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
Given a scenario where a session refresh was done with a removed
stream we would always add a removed stream to the outgoing SDP
even if one did not already exist.
This change makes it so that a removed stream is only placed into
the SDP if one already exists.
ASTERISK-28047
Change-Id: Ibb97d21cdeb87a8acae0c720861b0ff255708442
Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name
store IPv6 addresses without enclosing brackets. This causes some log
output to be confusing because it is difficult to separate the IPv6
address from a port specification.
* Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and
pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6
output.
* When a pj_sockaddr is not available, explicitly wrap IPv6 addresses
in brackets.
* When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure
to also set pjsip_rx_data.pkt_info.src_addr.
Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8
If in the initial sdp the caller doesn't include the line
a=rtcp-mux
Then asterisk shoud not include rtcp-mux in the response regardless
of rtcp-mux being enabled on the endpoint
ASTERISK-28007 #close
Change-Id: I58e9b9f40a139afc0da5de41906cc608fb62adc7
When a conference contained a mixture of audio/video and audio-only
users, a NOTICE message would pop up stating there are no joint
capabilities between streams. This happens because streams can never be
removed, but they can be in a REMOVED state. If we have the scenario
where user A joins with audio/video, user B joins with audio-only, and
user C joins with audio/video, then user A leaves, the message would
be triggered. That removed stream is still in the SDP, but Asterisk
would pass it through, causing it to be seen as a ulaw stream. A check
has been added for removed streams, setting their status to REMOVED when
handling negotiated SDPs.
Also addressed an issue where user A joins, then user B joins but does
not receive video until much later. Full frames were not being sent,
causing some PLI from the browser. Because the video was flowing in one
direction, the browser sets the SSRC to 1, but Asterisk was dropping the
PLI because of that. Added a check to see if the SSRC is 1 or not, which
sends full frames and allows video to flow between user A and user B.
This should only happen when dealing with PSFB or FUR, and in the case
of PSFB, only for PLI.
ASTERISK-27398
Change-Id: I26e7c6f101bc119549eeca406b5bcd25ad8ebc5e
Keep track if ICE candidates were in the SDP offer & only put them
in the corresponding SDP answer if the offer condaind ICE candidates
ASTERISK-27957 #close
Change-Id: Idf2597ee48e9a287e07aa4030bfa705430a13a92
When setting/appending the media id's to the bundle group attribute a '-1' was
being passed to the 'ast_str_set/append' function for the 'max_len' parameter.
This essentially capped the length of the string to what it was originally
allocated with. In this case 64 bytes.
This patch makes it so a '0' is passed as in for the 'max_len', which means
"no maximum length".
ASTERISK-27955 #close
Change-Id: Iec565df6600401d54a502854a53d19bb4cc34876
pjproject by default currently will follow media forked during an INVITE
on outbound calls if the To tag is different on a subsequent response as
that on an earlier response. We handle this correctly. There have
been reported cases where the To tag is the same but we still need to
follow the media. The pjproject patch in this commit adds the
capability to sip_inv and also adds the capability to control it at
runtime. The original "different tag" behavior was always controllable
at runtime but we never did anything with it and left it to default to
TRUE.
So, along with the pjproject patch, this commit adds options to both the
system and endpoint objects to control the two behaviors, and a small
logic change to session_inv_on_media_update in res_pjsip_session to
control the behavior at the endpoint level.
The default behavior for "different tags" remains the same at TRUE and
the default for "same tag" is FALSE.
Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6
ASTERISK-27936
Reported-by: Ross Beer
* Fix several instances where we were bumping a ref in the parameter and
then unrefing the object if it failed. The way the AST_VECTOR_APPEND()
and AST_VECTOR_REPLACE() macros are implemented means if it fails the new
value was never evaluated.
Change-Id: I2847872a455b11ea7e5b7ce697c0a455a1d0ac9a
It now appends the external IP address on the
o= line of the SDP packet. The decision was made to write
the numeric IP address as opposed to the RFC that states
the FQDN should be used if and when available. We believe
the usage of literal IP address will help avoid
potential problems.
ASTERISK-27614 #close
Change-Id: I84f3360f3606b8c4e8d161edb228799ec0b8a302
In handle_negotiated_sdp(), use session->active_media_state when
session->pending_media_state is empty. The 200's SDP should be fed into
handle_negotiated_sdp_session_media() together with the already negotiated
state, which is now in session->active_media_state instead. Only if both
the session's pending and active media are empty should
handle_negotiated_sdp() abort.
ASTERISK-27441
Change-Id: If0d5150ffe6f38d8a854831fef37942258d4629c
pjsip_distributor:
authenticate() creates a tdata and uses it to send a challenge or
failure response. When pjsip_endpt_send_response2() succeeds, it
automatically decrements the tdata ref count but when it fails, it
doesn't. Since we weren't checking for a return status, we weren't
decrementing the count ourselves on error and were therefore leaking
tdatas.
res_pjsip_session:
session_reinvite_on_rx_request wasn't decrementing the ref count
if an error happened while sending a 491 response.
pre_session_setup wasn't decrementing the ref count if
while sending an error after a pjsip_inv_verify_request failure.
res_pjsip:
ast_sip_send_response wasn't decrementing the ref count on error.
ASTERISK-27618
Reported By: Sandro Gauci
Change-Id: Iab33a6c7b6fba96148ed465b690ba8534ac961bf
pjproject does not have a function to reverse pjsip_inv_usage_init.
This means we need to ignore any calls to the functions once shutdown is
final.
ASTERISK-27571 #close
Change-Id: Ia550fcba563e2328f03162d79fb185f16b7c9b9d
This removes references that are no longer needed due to automatic
references created by module dependencies.
In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.
Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.
Still need to investigate dependencies among modules I cannot compile.
Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf
Some WebRTC clients can't handle renegotiation with the addition of
streams that include an offer to bundle. They instead expect the
newly added streams to already be bundled. This change does such a thing
if WebRTC support is enabled on an endpoint.
ASTERISK-27566
Change-Id: I7fe9b7ac35a2798627d9c2c8369129f407af6461
The pjsip_msg_find_hdr function can return NULL. This patch adds a check
when searching for the sequence header to make sure a NULL pointer is never
de-referenced.
Change-Id: I19af23aeeded65be016be92360e8cb7ffe51fad2
Fix instances of:
* Retreive
* Recieve
* other then
* different then
* Repeated words ("the the", "an an", "and and", etc).
* othterwise, teh
ASTERISK-24198 #close
Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31
When a connected line update is sent to an endpoint we do not request
a specific stream topology to be used. Previously this resulted in the
configured stream topology being used which may actually differ from the
currently negotiated topology. PJSIP is helpful in this regard in that
it will fill in any missing streams with removed ones. This results in
our own state not matching the SDP, though, and we do not apply the
negotiated SDP.
This change tweaks the code to use the actively negotiated stream
topology if it is present with a fallback to the configured one. This
results in the SDP and the state having matching information and the
world is happy.
ASTERISK*27397
Change-Id: I7a57117f0183479e6884b7bf3a53bb8c7464f604
This patch does three things associated with the initial incoming INVITE
request URI.
1) Add access to the full initial incoming INVITE request URI.
2) We were not setting DNID on incoming PJSIP channels. The DNID is the
user portion of the initial incoming INVITE Request-URI. The value is
accessed by reading CALLERID(dnid).
3) Fix CHANNEL(pjsip,target_uri) documentation.
* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).
* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.
* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.
* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.
ASTERISK-27478
Change-Id: I512e60d1f162395c946451becb37af3333337b33
A previous commit made it so when an invite session transitioned into a
disconnected state destruction of the Asterisk pjsip session object was
postponed until either a transport error occurred or the event timer
expired. However, if a call was rejected (for instance a 488) before the
session was fully established the event timer may not have been initiated,
or it was canceled without triggering either of the session finalizing states
mentioned above.
Really the only time destruction of the session should be delayed is when a
BYE is being transacted. This is because it's possible in some cases for the
session to be disconnected, but the BYE is still transacting.
This patch makes it so the session object always gets released (no more
memory leak) when the pjsip session is in a disconnected state. Except when
the method is a BYE. Then it waits until a transport error occurs or an event
timeout.
ASTERISK-27345 #close
Reported by: Corey Farrell
Change-Id: I1e724737b758c20ac76d19d3611e3d2876ae10ed
* Pre-initialize cloned media state vectors to final size to ensure
vector errors cannot happen later in the clone initialization.
* Release session_media on vector replace failure in
ast_sip_session_media_state_add.
* Release clone and media_state in ast_sip_session_refresh if we fail to
append to the stream topology, return an error.
Change-Id: Ib5ffc9b198683fa7e9bf166d74d30c1334c23acb
In WebRTC streams (or media tracks in their world) can be grouped
together using the mslabel. This informs the browser that each
should be synchronized with each other.
This change extends the stream API so this information can
be stored with streams. The PJSIP support has been extended
to use the mslabel to determine grouped streams and store
this association on the streams. Finally when creating the
SDP the group information is used to cause each media stream
to use the same mslabel.
ASTERISK-27379
Change-Id: Id6299aa031efe46254edbdc7973c534d54d641ad
PJSIP allows a domain name as external_media_address. This allows chan_pjsip to
be used behind a NAT with changing IP addresses. The IP address of that domain
is resolved to the c= line already. This change sets also the o= line to that
domain.
ASTERISK-27341 #close
Change-Id: I690163b6e762042ec38b3995aa5c9bea909d8ec4
Move ast_sip_add_usereqphone to be called after anonymization of URIs,
to prevent the user_eq_phone adding "user=phone" to URIs containing a
username that is not a phonenumber (RFC3261 19.1.1). An extra call to
ast_sip_add_usereqphone on the saved version before anonymization is
added to add user=phone" to the PAI.
ASTERISK-27047 #close
Change-Id: Ie5644bc66341b86dc08b1f7442210de2e6acdec6
res_pjsip and res_pjsip_session had circular references, preventing both
modules from shutting down.
* Move session supplement registration to res_pjsip.
* Use create internal functions for use by pjsip_message_filter.c.
ASTERISK-27306
Change-Id: Ifbd5c19ec848010111afeab2436f9699da06ba6b
For some scenarios when an outgoing call was made only a subset of the
configured codecs were offered. If the codecs being offered happened to
not have a codec supported by the phone then the call would fail.
For instance Alice and Bob both are configured in Asterisk for g722 and ulaw(
allow=!all,g722,ulaw). Alice's endpoint however only supports g722 while Bob's
only supports ulaw. When Alice calls Bob, Alice negotiates g722 fine with
Asterisk. But when Asterisk sends the outgoing offer to Bob it only contains
g722 and not both g722 and ulaw, so the call ends.
This patch makes it so all the audio codecs configured on the endpoint always
get sent, and not just a subset. However priority is given to those codecs that
are compatible with the "other side".
ASTERISK-27259 #close
Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696
When pruning a request to change the topology of a channel be
more intelligent about the resulting topology that is actually
used for SDP renegotiation.
In a case where a stream has not already been negotiated we
don't need to renegotiate and offer a declined stream. This can
occur if something in Asterisk (such as ConfBridge) requests
to add video to a PJSIP channel that has no video codecs configured.
In this case since the stream did not already exist we can safely
remove the stream from the requested topology, resulting in no
renegotiation occurring.
In a case where a renegotiation is requested with a codec that is
not supported we can reuse the formats of the existing stream if
it exists to ensure that the stream continues to flow, instead of
removing it.
Change-Id: I636540798d55922377318fe619c510fb6ed125fb
During a reinvite, if a remote endpoint error occurs and it returns a 500 the
session would end. This patch makes it so the session is not terminated, but
continues as it was.
The reason for this is because some endpoints may send non session terminating
"server errors" like a failed codec negotiation. So in this case instead of
ending the call it can hopefully continue. In the case of a real server error
the session is already "doomed", will be known soon enough and appropriately
ended by Asterisk later.
Change-Id: Ifeedae86b8cb44b92d52c79046522ec5f0aff1d5
When an INVITE came in with both audio and video streams but there
were no audio codecs defined for the endpoint, we weren't declining
the audio stream. Since it's usually the first/transport stream,
when the video stream was processed and tried to use the transport,
it was empty and caused a crash. We now decline the the stream if
there are no matching codecs so when the video stream is processed,
it's now the first/transport stream and processes normally.
Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692
Some endpoints do not like a stream being reused for a new
media stream. The frame/jitterbuffer can rely on underlying
attributes of the media stream in order to order the packets.
When a new stream takes its place without any notice the
buffer can get confused and the media ends up getting dropped.
This change uses the SSRC change to determine that a new source
is reusing an existing stream and then bridge_softmix renegotiates
each participant such that they see a new media stream. This
causes the frame/jitterbuffer to start fresh and work as expected.
ASTERISK-27277
Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07
There was an issue reported where an SDP received on a 183 Session
Progress message caused a crash because the pending streams had
already been processed when the OK was received. In that case the
pending topology was legitimately NULL. There was an assert for an
incorrect number of streams in the topology but not one for
topology being NULL. In any case, if you're not in dev-mode the
asserts don't do anything and since the scenario is legit, the
asserts weren't appropriate anyway.
* Changed several asserts to warning or debug messages and return
codes as appropriate.
ASTERISK-27264
Reported by: Daniel Heckl
Change-Id: I58daaa9d2938fa980857ab3ec41925ab5ff9c848
When a sip session is refreshed, the stream topology is looped
through, checking each stream for compatible formats. This would
cause a crash if the stream state was AST_STREAM_STATE_REMOVED,
since the formats would never be set for this stream, causing
a NULL value to be returned from ast_stream_get_formats. This
commit adds a check for streams with removed states.
Also removed a stray semicolon.
Change-Id: Ic86f8b65a4a26a60885b28b8b1a0b22e1b471d42
In 45744fc53, I mistakenly broke SDP media address rewriting by
misinterpreting which address was checked in the localnet comparison.
Instead of checking the remote peer address to decide whether we need
media address rewriting, we check our local media address: if it's
local, then we rewrite. This feels awkward, but works and even made
directmedia work properly if you set local_net. (For the record: for
local peers, the SDP media rewrite code is not called, so the
comparison does no harm there.)
ASTERISK-27248 #close
Change-Id: I566be1c33f4d0a689567d451ed46bab9c3861d4f
In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was
confusion about whether the transport_state->localnet ACL has ALLOW or
DENY semantics.
For the record: the localnet has DENY semantics, meaning that "not in
the list" means ALLOW, and the local nets are in the list.
Therefore, checks like this look wrong, but are right:
/* See if where we are sending this request is local or not, and if
not that we can get a Contact URI to modify */
if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) {
ast_debug(5, "Request is being sent to local address, "
"skipping NAT manipulation\n");
(In the list == localnet == DENY == skip NAT manipulation.)
And conversely, other checks that looked right, were wrong.
This change adds two macro's to reduce the confusion and uses those
instead:
ast_sip_transport_is_nonlocal(transport_state, addr)
ast_sip_transport_is_local(transport_state, addr)
ASTERISK-27248 #close
Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934
Stream names within Asterisk can have meaning so when an externally
initiated renegotiation occurs we need to preserve the name of
the stream if it already exists.
Change-Id: I29f50d0cc7f3238287d6d647777e76e1bdf8c596
If an SDP answer hasn't been sent yet, it's legal to change it.
This is required for PJSIP_DTMF_MODE to work correctly, and can
also have use in the future for updating codecs too.
ASTERISK-27209 #close
Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1
If a previously active stream is declined we could crash because the
channel's thread is still using the stream while we are updating the
topology in the serializer thread.
* Defer removing any declined stream's handler until we have blocked the
channel's thread with the channel lock.
ASTERISK-27212
Change-Id: I50e1d3ef26f8e41948f4c411ee329aa3b960a420
A change was made long ago where the session was kept around
until the underlying INVITE session had been destroyed. This
had the side effect of also keeping the underlying media resources
around for this time as well.
This change ensures that when we are told to terminate the
session we immediately release any media sessions associated
with it.
ASTERISK-27110
Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82
Currently, the handling of the msid attribute is not quite right. According to
the spec the msid's between the offer/answer are not dependent upon one another.
Meaning the same msid's given in an offer do not have to be returned in the
answer for a given stream. And they probably shouldn't be (copied/reused) since
this can potentially cause some browser side confusion.
This patch generates new msids when both an offer and answer are sent from
Asterisk. However, Asterisk does reuse the original msid it sent out for a
reinvite. Also audio+video streams are paired together by sharing the same
stream id, but a different track id.
ASTERISK-27179 #close
Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643
The "external_media_address" option on transports is now
resolved using dnsmgr. This allows it to be automatically
refreshed regularly if refreshes are enabled in dnsmgr.
If the system is using a dynamic IP address a dynamic DNS
hostname can be provided to keep the IP address up to
date.
Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2
This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis
ASTERISK-27085 #close
Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612
This change does a few things to improve packet loss and renegotiation:
1. On outgoing RTP streams we will now properly reflect out of order
packets and packet loss in the sequence number. This allows the
remote jitterbuffer to better reorder things.
2. Video updates can now be discarded for a period of time
after one has been sent to prevent flooding of clients.
3. For declined and removed streams we will now release any
media session resources associated with them. This was not
previously done and caused an issue where old state was being
used for a new stream.
4. RTP bundling was not actually removing bundled RTP instances
from the parent. This has been resolved by removing based on
the RTP instance itself and not the SSRC.
5. The code did not properly handle explicitly unbundling an
RTP instance from its parent. This now works as expected.
ASTERISK-27143
Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45
This patch creates a new configuration option called "webrtc". When enabled it
defaults and enables the following options that are needed in order for webrtc
to work in Asterisk:
rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
When "webrtc" is enabled, this patch also parses the "msid" media level
attribute from an SDP. It will also appropriately add it onto the outgoing
session when applicable.
Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent.
ASTERISK-27119 #close
Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd
BUNDLE is a specification used in WebRTC to allow multiple
streams to use the same underlying transport. This reduces
the number of ICE and DTLS negotiations that has to occur
to 1 normally.
This change implements this by adding support for it to
the RTP SDP module in PJSIP. BUNDLE can be turned on using
the "bundle" option and on an offer we will offer to
bundle streams together. On an answer we will accept any
bundle groups provided. Once accepted each stream is bundled
to another RTP instance for transport.
For the res_rtp_asterisk changes the ability to bundle
an RTP instance to another based on the SSRC received
from the remote side has been added. For outgoing traffic
if an RTP instance is bundled to another we will use the
other RTP instance for any transport related things. For
incoming traffic received from the transport instance we
look up the correct instance based on the SSRC and use it
for any non-transport related data.
ASTERISK-27118
Change-Id: I96c0920b9f9aca7382256484765a239017973c11
The stream topology (list of streams and order) is now stored with the
configured PJSIP endpoints and used during the negotiation process.
Media negotiation state information has been changed to be stored
in a separate object. Two of these objects exist at any one time
on a session. The active media state information is what was previously
negotiated and the pending media state information is what the
media state will become if negotiation succeeds. Streams and other
state information is stored in this object using the index (or
position) of each individual stream for easy lookup.
The ability for a media type handler to specify a callback for
writing has been added as well as the ability to add file
descriptors with a callback which is invoked when data is available
to be read on them. This allows media logic to live outside of
the chan_pjsip module.
Direct media has been changed so that only the first audio and
video stream are directly connected. In the future once the RTP
engine glue API has been updated to know about streams each individual
stream can be directly connected as appropriate.
Media negotiation itself will currently answer all the provided streams
on an offer within configured limits and on an offer will use the
topology created as a result of the disallow/allow codec lines.
If a stream has been removed or declined we will now mark it as such
within the resulting SDP.
Applications can now also request that the stream topology change.
If we are told to do so we will limit any provided formats to the ones
configured on the endpoint and send a re-invite with the new topology.
Two new configuration options have also been added to PJSIP endpoints:
max_audio_streams: determines the maximum number of audio streams to
offer/accept from an endpoint. Defaults to 1.
max_video_streams: determines the maximum number of video streams to
offer/accept from an endpoint. Defaults to 1.
ASTERISK-27076
Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7
There was a typo introduced in commit 776ffd77 which was preventing
the transport's external media address from being used.
ASTERISK-27024 #close
Reported-by: Christopher van de Sande
patches:
patch.diff submitted by Florian Floimair (license 6892)
Change-Id: I7ec617171eaa2d86d2680b00cf37d5088adafc27
When doing an attended transfer it's possible for the transferer, after
receiving an accepted response from Asterisk, to send a BYE to Asterisk,
which can then be processed before Asterisk has time to start and/or
complete the transfer process. This of course causes the transfer to not
complete successfully, thus dropping the call.
This patch makes it so any BYEs received from the transferer, after the REFER,
that initiate a session end are deferred until the transfer is complete. This
allows the channel that would have otherwise been hung up by Asterisk to
remain available throughout the transfer process.
ASTERISK-27053 #close
Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a
This introduces the ability for PJSIP code to specify filtering flags
when retrieving PJSIP contacts. The first flag for use causes the
query code to only retrieve contacts that are not unreachable. This
change has been leveraged by both the Dial() process and the
PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt
calls to contacts which are not unreachable.
ASTERISK-26281
Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c
Retransmissions of an initial INVITE could be queued in the serializer
before we have processed the first INVITE message. If the first INVITE
message doesn't get completely processed before the retransmissions are
seen then we could try to setup the same call from the retransmissions. A
symptom of this is seeing a (key exists) message associated with an
INVITE. An earlier change attempted to address this kind of problem by
calculating a distributor serializer to use for unassociated messages.
Part of that change also made incoming calls keep using that distributor
serializer. (ASTERISK-26088) However, some leftover code was still
deferring the INVITE processing to the session's serializer even though we
were already in that serializer. This not only is unnecessary but would
cause the same call resetup problem.
* Removed the code to defer processing the initial INVITE to the session's
serializer because we are already running in that serializer.
ASTERISK-26998 #close
Change-Id: I1e822d82dcc650e508bc2d40d545d5de4f3421f6
If you use ast_request to create a PJSIP channel but then hang it
up without causing a transaction to be sent, the session will
never be destroyed. This is due ot the fact that it's pjproject
that triggers the session cleanup when the transaction ends.
app_chanisavail was doing this to get more granular channel state
and it's also possible for this to happen via ARI.
* ast_sip_session_terminate was modified to explicitly call the
cleanup tasks and unreference session if the invite state is NULL
AND invite_tsx is NULL (meaning we never sent a transaction).
* chan_pjsip/hangup was modified to bump session before it calls
ast_sip_session_terminate to insure that session stays valid
while it does its own cleanup.
* Added test events to session_destructor for a future testsuite
test.
ASTERISK-26908 #close
Reported-by: Richard Mudgett
Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9
If ICE is enabled and a STUN server does not respond then we will block
until we give up on the STUN response. This will take nine seconds. In
the mean time the peer that sent the INVITE will send retransmissions.
* Restructure res_pjsip_session.c:new_invite() to send a 100 Trying out
earlier to prevent these retransmissions.
ASTERISK-26890
Change-Id: Ie3fc611e53a0eff6586ad55e4aacad81cf6319a8
* Restructure ast_sip_session_alloc() to need less cleanup on off nominal
error paths.
* Made ast_sip_session_alloc() and ast_sip_session_create_outgoing() avoid
unnecessary ref manipulation to return a session. This is faster than
calling a function. That function may do logging of the ref changes with
REF_DEBUG enabled.
Change-Id: I2a0affc4be51013d3f0485782c96b8fee3ddb00a
It is perfectly acceptable for a BYE to be sent on a disconnected
session. This occurs when we respond to a challenge to the BYE
for authentication credentials.
ASTERISK-26363
Change-Id: I6ef0ddece812fea6665a1dd2549ef44fb9d90045
Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched
destinations) as currently provided by chan_sip is missing from res_pjsip.
This patch adds a new endpoint attribute (allow_overlap) [defaults to yes]
which when set to yes enables 484 responses to partial destination
matches rather than the current 404.
ASTERISK-26864
Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6
If local_net is not defined on a transport, transport_state->localnet
will be NULL. ast_apply_ha will, be default, return AST_SENSE_ALLOW in
this case, causing the external_media_address, if set, to be skipped.
This patch causes us to only check if we are sending within a network if
local_net is defined.
ASTERISK-26879 #close
Change-Id: Ib661c31a954cabc9c99f1f25c9c9a5c5b82cbbfb
A new transport parameter 'symmetric_transport' has been added.
When a request from a dynamic contact comes in on a transport with
this option set to 'yes', the transport name will be saved and used
for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE.
It's saved as a contact uri parameter named 'x-ast-txp' and will
display with the contact uri in CLI, AMI, and ARI output. On the
outgoing request, if a transport wasn't explicitly set on the
endpoint AND the request URI is not a hostname, the saved transport
will be used and the 'x-ast-txp' parameter stripped from the
outgoing packet.
* config_transport was modified to accept and store the new parameter.
* config_transport/transport_apply was updated to store the transport
name in the pjsip_transport->info field using the pjsip_transport->pool
on UDP transports.
* A 'multihomed_on_rx_message' function was added to
pjsip_message_ip_updater that, for incoming requests, retrieves the
transport name from pjsip_transport->info and retrieves the transport.
If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter
containing the transport name is added to the incoming Contact header.
* An 'ast_sip_get_transport_name' function was added to res_pjsip.
It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a
transport name if endpoint->transport is set or if there's an
'x-ast-txp' parameter on the uri and the uri host is an ipv4 or
ipv6 address. Otherwise it returns NULL.
* An 'ast_sip_dlg_set_transport' function was added to res_pjsip
which takes an ast_sip_endpoint, a pjsip_dialog, and an optional
pjsip_tpselector. It calls ast_sip_get_transport_name() and if
a non-NULL is returned, sets the selector and sets the transport
on the dialog. If a selector was passed in, it's updated.
* res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas
were modified to call ast_sip_dlg_set_transport() instead of their
original logic.
* res_pjsip/create_out_of_dialog_request was modified to call
ast_sip_get_transport_name() and pjsip_tx_data_set_transport()
instead of its original logic.
* Existing transport logic was removed from endpt_send_request
since that can only be called after a create_out_of_dialog_request.
* res_pjsip/ast_sip_create_rdata was converted to a wrapper around
a new 'ast_sip_create_rdata_with_contact' function which allows
a contact_uri to be specified in addition to the existing
parameters. (See below)
* res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated
since all it did was transport selection and that is now done in
ast_sip_create_dialog_uac and ast_sip_create_dialog_uas.
* 'contact_uri' was added to subscription_persistence. This was
necessary because although the parsed rdata contact header has the
x-ast-txp parameter added (if appropriate),
subscription_persistence_update stores the raw packet which
doesn't have it. subscription_persistence_recreate was then
updated to call ast_sip_create_rdata_with_contact with the
persisted contact_uri so the recreated subscription has the
correct transport info to send the NOTIFYs.
* res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since
all it did was transport selection and that is now done in
ast_sip_create_dialog_uac.
* pjsip_message_ip_updater/multihomed_on_tx_message was updated
to remove all traces of the x-ast-txp parameter from the
outgoing headers.
NOTE: This change does NOT modify the behavior of permanent
contacts specified on an aor. To do so would require that the
permanent contact's contact uri be updated with the x-ast-txp
parameter and the aor sorcery object updated. If we need to
persue this, we need to think about cloning permanent contacts into
the same store as the dynamic ones on an aor load so they can be
updated without disturbing the originally configured value.
You CAN add the x-ast-txp parameter to a permanent contact's uri
but it would be much simpler to just set endpoint->transport.
Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f
This feature was available in the SIP channel driver chan_sip. For example,
Asterisk is the outbound proxy and has to handle all SIP-URIs, even domains not
local to Asterisk. In that case, SIPDOMAIN is used in the Dialplan, to detect
and dial remote SIP-URIs. This change here sets the SIP destination domain of
an inbound call (SIPDOMAIN) in the SIP channel driver res_pjsip as well.
ASTERISK-26670 #close
Change-Id: I27c880dc404a3c1c6792e1ba3545475339577243
Joshua C. Colp
DanielYK
George Joseph
Torrey Searle
Sungtae Kim
Kevin Harwell
Sean Bright
Stas Kobzar
Sungtae Kim
Corey Farrell
Nick French
Joshua Colp
Ben Ford
Richard Mudgett
Chris-Savinovich
lvl
Jenkins2
Alexander Traud
Daniel Tryba
Walter Doekes
Mark Michelson
Yasin CANER
Richard Begg
Matt Jordan