ASTERISK-26776 #close

Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2

POSIX does not require getprotobyname() to be thread safe and some
implementations use static memory which causes issues when multiple
threads are used.

Further, our usage of it today is just to ultimately get IPPROTO_TCP
for calls to setsockopt(). So instead we just use IPPROTO_TCP directly.

Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48

We are currently passing in the capacity of the read buffer instead of the
number of bytes that we actually read off the wire.

Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36

The queue_stasis_data structure contains various mutable fields that require
appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and
'caller_uniqueid' fields need to be locked when read from or written to.

Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088

ASTERISK-26846 #close

Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639

Thanks to Chris Howard for pointing this out on the wiki.

Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705

struct ast_rtcp does not define the dtls member if SRTP is not enabled.

ASTERISK-26732

Change-Id: Id15ea212e04490e012f2cf4a56818b4dd948875e

We were inadvertenly referencing the cos_video option to determine if we
should set the tos_audio and cos_audio value on the RTP instance.

Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0

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

Currently a wildcard address is used for the local RTP socket, which
will not always result in the same address as used by the SIP socket
(e.g. if explicit transport addresses are configured).
Use the transport's host address when binding new local RTP sockets if
available.

ASTERISK-26851

Change-Id: I098c29c9d1f79a4f970d72ba894874ac75954f1a

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 change removes an assumption that when DTLS is stopped
an RTCP session will be present on the RTP session. This is not
always the case.

ASTERISK-26732

Change-Id: Ib9f7c09ce0b005efe362dbcc8795202b18f94611

This commit adds support for RFC 5761: Multiplexing RTP Data and Control
Packets on a Single Port. Specifically, it enables the feature when
using chan_pjsip.

A new option, "rtcp_mux" has been added to endpoint configuration in
pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with
whatever it communicates with. Asterisk follows the rules set forth in
RFC 5761 with regards to falling back to standard RTCP behavior if the
far end does not indicate support for rtcp-mux.

The lion's share of the changes in this commit are in
res_rtp_asterisk.c. This is because it was pretty much hard wired to
have an RTP and an RTCP transport. The strategy used here is that when
rtcp-mux is enabled, the current RTCP transport and its trappings (such
as DTLS SSL session) are freed, and the RTCP session instead just
mooches off the RTP session. This leads to a lot of specialized if
statements throughout.

ASTERISK-26732 #close
Reported by Dan Jenkins

Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5

When transfering to a URI without an extension, ensure that the
s extension of the dialplan is entered

ASTERISK-26869 #close

Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525

A caller can leave the Queue() application after being bridged with a
member in a few ways:

  * Caller or member hangup
  * Caller is transferred somewhere else (blind or atx)
  * Caller is externally redirected elsewhere

The first 2 scenarios are currently handled by subscribing to stasis
messages, but the 3rd is not explicitly covered. If a caller is
redirected away from the Queue() application, the member who was last
bridged with that caller will remain in an "In use" state until the
caller hangs up.

This patch adds handling of the caller leaving the queue via
redirection. We monitor the caller-member bridge, and if the caller is
the one that leaves, we treat it the same as we would a caller hangup.

ASTERISK-26400 #close
Reported by: Etienne Lessard

Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334

This change ensures that if no header_match option is set on an
identify an error message is not output stating the option is set
to an invalid value.

ASTERISK-26863

Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a

This patch adds a new features to the endpoint identifier module,
'match_header'. When set, inbound requests are matched by a provided SIP
header: value pair. This option works in conjunction with the existing
'match' configuration option, such that if any 'match*' attribute
matches an inbound request, the request is associated with the specified
endpoint.

Since this module now identifies by more than just IP address,
appropriate renaming of the module and/or variables can be done in a
non-release branch.

ASTERISK-26863 #close

Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453
(cherry picked from commit 30f52d79)