Added the stun_blacklist option to rtp.conf. Some multihomed servers have
IP interfaces that cannot reach the STUN server specified by stunaddr.
Blacklist those interface subnets from trying to send a STUN packet to
find the external IP address. Attempting to send the STUN packet
needlessly delays processing incoming and outgoing SIP INVITEs because we
will wait for a response that can never come until we give up on the
response. Multiple subnets may be listed.
ASTERISK-26890 #close
Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342
* create_rtp(): Eliminate use of deprecated transport struct member. That
member and several others in the transport structure were deprecated
because of an infinite loop created when using realtime configuration.
See 2451d4e455
ASTERISK-26851
Change-Id: I0533aa13c9ce3c6cc394e0fd2b5bf1cd1b2ef3bc
This change adds database tables for the PUBLISH support so it
can be configured using realtime. A minor fix to the
res_pjsip_publish_asterisk module was done so that it read the
sorcery configuration from the correct section. Finally the
sample configuration files have been updated.
ASTERISK-26928
Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952
* create_rtp(): Fix unexpected alteration of global address_rtp if a
transport is bound to an address.
* create_rtp(): Fix use of uninitialized memory if the endpoint RTP media
address is invalid or the transport has an invalid address.
ASTERISK-26851
Change-Id: Icde42e65164a88913cb5c2601b285eebcff397b7
We were leaking a transport ref in multihomed_on_rx_message() which
resulted in the FRACK about excessive ref counts.
ASTERISK-26916 #close
Change-Id: I7a96658a9614a060565bb9ad51cb1c9c11ee145f
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
Two new parameters have been added to the pjsip config wizard.
* Setting 'sends_line_with_registrations' to true will cause the wizard
to skip the creation of an identify object to match incoming request
to the endpoint and instead add the line and endpoint parameters to
the outbound registration object.
* Setting 'outbound_proxy' is a shortcut for adding individual
endpoint/outbound_proxy, aor/outbound_proxy and
registration/outbound_proxy parameters.
Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0
There doesn't appear to be any reason that we are chdir'ing in
moh_scan_files, and in the event of an Asterisk crash, the core files
may not get written because we have changed into a read-only directory.
ASTERISK-23996 #close
Reported by: Walter Doekes
Change-Id: Iac806dce01b3335963fbd62d4b4da9a65c614354
If a read error occurs, we immediately attempt a reconnect without any
delay. Instead, let's sleep and backoff up to 60 seconds before we try
again.
ASTERISK-24712 #close
Reported by: Matthias Urlichs
Change-Id: I6fe10ef4734837727437beab715e336777f13f48
If any errors occur during the TLS connection setup, we currently dump a
fairly generic error message. So instead we try to pull in something
useful from OpenSSL to report instead.
ASTERISK-24712
Reported by: Matthias Urlichs
Change-Id: I288500991a9681f447d92913b11fedaf426087f4
SSL_connect returns non-zero for both success and some error conditions
so simply negating is inadequate.
Change-Id: Ifbf882896e598703b6c615407fa456d3199f95b1
The only remaining reference to the endpoint is in the endpoints
container, and because it is unlinked in ast_endpoint_shutdown, we don't
have to explicitly cleanup the endpoint ourselves.
Change-Id: I912a2692e52d3e2ed445b32d8ae3f9004bc2f2e8
chan_sip sets the hangup cause code to AST_CAUSE_REQUESTED_CHAN_UNAVAIL
(44) when a channel is hung up due to an RTP timeout. So do the same
when it happens with PJSIP for parity.
Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8
The documentation for JABBER_STATUS (and the deprecated JabberStatus
app) indicate that a return value of 7 indicates that the specified
buddy was not in the roster. It also indicates that you can specify a
"bare" JID (one without a resource). Unfortunately the actual behavior
does not match the documented behavior.
Assuming that our roster includes the buddy online and available
"valid@example.org/Valid" and does *not* include the buddy
"invalid@example.org", the JABBER_STATUS() function returns the
following before this patch:
+------------------------------+------------+--------------------------+
| Buddy | Status | Result |
+------------------------------+------------+--------------------------+
| valid@example.org | Online | 7 (Not in roster) |
| valid@example.org/Valid | Online | 1 (Online) |
| valid@example.org/Invalid | N/A | 7 (Not in roster) |
| invalid@example.org | N/A | Error logged, no return |
| invalid@example.org/Valid | N/A | Error logged, no return |
+------------------------------+------------+--------------------------+
And after this patch:
+------------------------------+------------+--------------------------+
| Buddy | Status | Result |
+------------------------------+------------+--------------------------+
| valid@example.org | Online | 1 (Online) |
| valid@example.org/Valid | Online | 1 (Online) |
| valid@example.org/Invalid | N/A | 6 (Offline) |
| invalid@example.org | N/A | 7 (Not in roster) |
| invalid@example.org/Valid | N/A | 7 (Not in roster) |
+------------------------------+------------+--------------------------+
This brings the behavior in line with the documentation.
ASTERISK-23510 #close
Reported by: Anthony Critelli
Change-Id: I9c3241035363ef4a6bdc21fabfd8ffcd9ec657bf
If we never establish a connection to our Jabber server, iksemel never sets up
its internal transport pointer, so attempting to send a message dereferences a
NULL pointer and causes a crash.
ASTERISK-21855 #close
Reported by: Jeremy Kister
Change-Id: I204a568894e4a53ab929783ecc594a000f04d79c
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
Rather than hard-coding UDP, allow consumers of the HEP API to specify
which protocol is in use. Update the PJSIP provider to pass in the
current protocol type.
ASTERISK-26850 #close
Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978
We aren't validating that the URI we just parsed is a SIP/SIPS one before
trying to access the user, host, and port members of a possibly uninitialized
structure.
Also update the MessageSend documentation to indicate what 'from' formats are
accepted.
ASTERISK-26484 #close
Reported by: Vinod Dharashive
Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30
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
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
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
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 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
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
When transfering to a URI without an extension, ensure that the
s extension of the dialplan is entered
ASTERISK-26869 #close
Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525
