We use the transport of the Contact header of the remote UAC to decide
which of our own Contact addresses we should use when replying to a
SUBSCRIBE or sending a presence NOTIFY.
If TLS is not enabled on a Sofia profile, then the TLS Contacts for
that profile are NULL. Unfortunately we were using these NULL values
uncritically when the remote UAC sent us a Contact header with a TLS
transport and our own Sofia profile did not have TLS enabled.
With this commit we fall back to our TCP Contact address when the
remote Contact is TLS and our Sofia profile does not have TLS enabled.
ADTRAN Total Access devices do not support sending the rport parameter in
the Via header. This allows us to detect the device and force rport when
using the "safe" parameter, enabling the device to be used behind NAT.
FS-6823 #resolve
When all-reg-options-ping is enabled, this adds a new custom event to mod_sofia
(sofia::sip_user_state), which is fired when a client stops responding to such
ping packets (or when it is reachable again).
Add two needed new columns to the sip_registrations table:
- ping_status, which is "Reachable" or "Unreachable" depending on the client
status;
- ping_count, which tracks the number of ping responses received and is used
to provide some kind of hysteresis to avoid firing the event in case of
transitory network failures.
Then ping_count is checked against two threshold values, sip-user-ping-min
and sip-user-ping-max in a similar fashion as the ping-{max,min} options for
the gateways. These two values are configurable in the profile's xml
configuration file.
Also, if unregister-on-options-fail is enabled, the client is unregistered
based on the number of OPTIONS failure which is also checked against the
sip-user-ping-{min,max} values.
VARIABLE: bypass_media_sdp_filter
Can be set globally or per leg on the inbound side of a bypass_media bridge.
VALID FILTERS:
remove(): Removes the specified codec if it exists in the SDP.
only(): Removes all codecs besides the one specified (providing that it exists in the sdp) (will not remove telephone-event))
EXAMPLE 1 (remove everything leaving only g729):
<action application="set" data="bypass_media_sdp_filter=only(g729)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
EXAMPLE 2 (remove everything leaving only g729 and also remove dtmf):
<action application="set" data="bypass_media_sdp_filter=only(g729)|remove(telephone-event)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
EXAMPLE 3 (remove alaw and speex):
<action application="set" data="bypass_media_sdp_filter=remove(pcma)|remove(speex)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
This patch allows conference flags to be set dynamically from the
dial plan by either passing them to the conference application in
the +flags{ } string or by setting the "conference_flags" dial plan
variable.
The +flags{ } string is currently used to set *user* flags only.
This patch changes this by allowing the +flags{ } string to contain
conference related flags as well (for example wait_mod). It shouldn't
be a problem to pass both types of flags via +flags{ } as long as
the user and conference flag names are kept unique.
FS-5099 #resolve
