Although it's perfectly legal to place multiple SIP messages in the same packet,
it can cause problems because the Linux default is to enable Path MTU Discovery
which sets the Don't Fragment bit on the packets. If adding a second message to
the packet causes the MTU to be exceeded, and the destination isn't equipped to
send a FRAGMENTATION NEEDED response to a large packet, the packet will just be
dropped.
We can't specifically tell the stack to send only 1 message per packet, but we
can turn on TCP_NODELAY when we create the transport. This will at least tell
the stack to send packets as soon as possible.
ASTERISK-26005 #close
Reported-by: Ross Beer
Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd
The Location headers returned by:
* /bridges/{bridgeId}/play
* /bridges/{bridgeId}/record
* /channels/{channelId}/play
* /channels/{channelId}/record
Did not have the '/ari' prefix, and in the case of the 'play' resources, were
using 'playback' instead of 'playbacks.'
Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c
At one point in time, it seemed like a good idea to use the Asterisk
channel name as the HEP correlation UUID. In particular, it felt like
this would be a useful identifier to tie PJSIP messages and RTCP
messages together, along with whatever other data we may eventually send
to Homer. This also had the benefit of keeping the correlation UUID
channel technology agnostic.
In practice, it isn't as useful as hoped, for two reasons:
1) The first INVITE request received doesn't have a channel. As a
result, there is always an 'odd message out', leading it to be
potentially uncorrelated in Homer.
2) Other systems sending capture packets (Kamailio) use the SIP Call-ID.
This causes RTCP information to be uncorrelated to the SIP message
traffic seen by those capture nodes.
In order to support both (in case someone is trying to use res_hep_rtcp
with a non-PJSIP channel), this patch adds a new option, uuid_type, with
two valid values - 'call-id' and 'channel'. The uuid_type option is used
by a module to determine the preferred UUID type. When available, that
source of a correlation UUID is used; when not, the more readily available
source is used.
For res_hep_pjsip:
- uuid_type = call-id: the module uses the SIP Call-ID header value
- uuid_type = channel: the module uses the channel name if available,
falling back to SIP Call-ID if not
For res_hep_rtcp:
- uuid_type = call-id: the module uses the SIP Call-ID header if the
channel type is PJSIP and we have a channel,
falling back to the Stasis event provided
channel name if not
- uuid_type = channel: the module uses the channel name
ASTERISK-25352 #close
Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c
A minimal configuration does not need to explicitly spell out the
directories. The built-in defaults will do just fine. In many cases
they are wrong.
Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
Note the default of remmed-out options. To clarify that those values are
not the defaults.
Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* No need to set language in a miniml configuration. 'en' will do just
fine.
* It would be useful to have an example of setting it to a different
language.
* Setting the documentation language explicitly is likewise not
required. Setting it to a different value is not common. At least
until there is a set of translated documentation.
Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
FollowMe with the option a records the name of the caller and plays it
to the callee. However it has failed to clean up that recorded file
as it tried to delete the file name without the '.sln' extension.
ASTERISK-26008 #close
Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
In 13.9.0, there was an issue where PJSIP contacts added to an AOR would
be deleted at seemingly random times.
One reason this was happening was because of an operation to retrieve
the contacts whose expiration time was less than or equal to the current
time. When retrieving existing contacts, the contact's expiration time
and the current time were converted from a string to a float, and those
two floats were compared.
On some systems, including mine, this conversion was horribly off. For
instance, I could regularly see the string "1463079214" get converted
into 1463079168.000000. When switching from using a float to using a
double, the conversion was as expected.
Why was the conversion to float off? My best guess is that the
conversion to float was attempting to store the entire value in the 23
bit significand of the IEEE-754 floating point number. In particular, if
you take only the 23 most significant bits of 1463079214, you get the
messed up 1463079168 that we were seeing in the conversion. It likely
was possible to get a more precise value by composing the number using
an exponent, but the conversion did not work that way. With a double,
you have a 52 bit significand, allowing the entire value to fit there,
and thereby allowing an accurate conversion.
ASTERISK-26007 #close
Reported by Greg Siemon
Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070
There are two types of SIP URIs indicating a secure transport:
* sips:user@example.org
* sip:user@example.org;transport=tls
When using a sips URI, Asterisk checks incoming INVITEs and answers from
the other side for sips URIs, and rejects the packet if there are only
sip URIs. So Asterisk should only generate a sips Contact URI if the
other side supports it.
This patch makes Asterisk generate either a sip or sips Contact URI
depending on the format of the server URI.
If you want a sip URI, use:
server_uri=sip:example.org\;transport=tls
If you want a sips URI, use:
server_uri=sips:example.org
ASTERISK-25990 #close
Reported-by: Sebastian Damm
Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2
During refactoring of this support the addition of
the PID to messages was removed. This change adds it
back in.
ASTERISK-25538 #close
Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36
When running on a system that does not support or use AST_UNDEFINED_SANITIZER
or AST_LEAK_SANITIZER, the configure script would incorrectly set those
constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would
cause menuselect to error out, complaining that a blank value is not a
valid option. This patch corrects the issue by setting the value to 0 if
the options that those constants enable/disable is not found.
Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba
From the issue reporter:
"res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of
the timestamp, the source address, the source port, a server UUID that is
calculated at startup, and the authentication realm.
Rather than caching nonces that we create, we instead attempt to re-calculate
the nonce when receiving an incoming request with authentication. We then
compare the re-calculated nonce to the incoming nonce, and if they don't match,
then authentication has failed early.
The problem is that it is possible, especially when using TCP, to receive two
requests from the same endpoint but have differing source ports for those
requests. Asterisk itself commonly will use different source ports for
outbound TCP requests."
This patch removes the source port dependency when building the nonce.
ASTERISK-25978 #close
Change-Id: I871b5f4adce102df1c4988066283095ec509dffe
The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2.
SSL is not allowed. So, even if you specify "sslv3" for a transport method,
it's silently ignored and one of the TLS protocols is used. This was a new
behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that
we never caught.
Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default().
This tells pjproject to set the socket protocol to match the method.
ASTERISK-26004 #close
Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078
This patch allows for having app_confbridge register the name of the
conference as an extension into a specific context, similar to
regcontext for chan_sip. This variant is not quite as involved as the
one in chan_sip and doesn't allow for multiple contexts or custom
extensions, you can only specify the context and the conference name
will always be used as the extension to register.
ASTERISK-25989 #close
Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f
The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set.
The python bindings are now built only if TEST_FRAMEWORK is set and a
python development package is installed.
libresample was also disabled.
ASTERISK-25993 #close
Reported-by: Joshua Colp
Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03
The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_*
and res_pjsip_registrar modules should load ASAP
to avoid "No matching endpoint found" for legitimate endpoint.
ASTERISK-25994
Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b
ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail.
ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail.
These additions should be also in stasis_endpoints
to include in command "manager show event ContactStatus"
Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a
It is possible for the nativeformats of a channel to change
throughout its lifetime. As a result a user of it needs to either
ensure the channel is locked when accessing the formats or keep
a reference to the nativeformats themselves.
This change fixes the file playback support so it keeps a
reference to the nativeformats when accessing things.
ASTERISK-25998 #close
Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915
This patch modified pjsip_options to retrieve only
permament contacts for aor if the qualify_frequency is > 0
and persisted contacts if the qualify_frequency is > 0.
This patch also fixed a bug in res_sorcery_astdb.
res_sorcery_astdb doesn't save object data retrived from astdb.
ASTERISK-25826
Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05
With the old SIP module AMI sends PeerStatus event on every
successfully REGISTER requests, ie, on start registration,
update registration and stop registration.
With PJSIP AMI sends ContactStatus only when status is changed.
Regarding registration:
on start registration - Created
on stop registration - Removed
but on update registration nothing
This patch added contact.updated event.
ASTERISK-25904
Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f
