1
0
Fork 0
mirror of https://gitlab.gnome.org/GNOME/calls.git synced 2024-11-06 00:21:19 +00:00
Commit graph

135 commits

Author SHA1 Message Date
Evangelos Ribeiro Tzaras
a44c265903 sip: remove FOR_TESTING ifdef 2021-04-06 16:55:33 +00:00
Evangelos Ribeiro Tzaras
4a264252a7 plugins: build as shared library instead of shared module 2021-04-06 16:55:33 +00:00
Evangelos Ribeiro Tzaras
7d69d78b70 origin: Add country-code property
And add a binding in CallsManager for the default origin
2021-04-06 14:27:26 +00:00
Evangelos Ribeiro Tzaras
75d32d0924 sip: Allow controlling automatic account loading via environment 2021-04-05 06:13:44 +00:00
Mohammed Sadiq
c30a41ffa9 Let calls-call be an abstract class
And adapt to changes.

A calls-mm-call IS-A calls-call (and so on)
2021-04-05 09:38:03 +05:30
Evangelos Ribeiro Tzaras
e6b730b805 sip: pipeline: clean up in finalize () 2021-04-03 00:46:29 +02:00
Evangelos Ribeiro Tzaras
71cbc5c636 sip: provider: Fall back to reasonable values for local-port property 2021-04-03 00:46:29 +02:00
Evangelos Ribeiro Tzaras
5a7c22c80f sip: provider: use g_get_user_config_dir () for account config 2021-04-03 00:46:29 +02:00
Evangelos Ribeiro Tzaras
840ffa4653 sip: do not auto load accounts when running tests
because `calls_sip_provider_load_accounts ()` looks at the home folder
for a configuration file.
2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
19e7f8f119 sip-media: enable echo cancellation
by setting "filter.want" to "echo-cancel" for the pulsesink and pulsesrc
GStElement's.
2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
8a6f1bb203 sip: fix infinite ringtone loop
by making sure the call-added signal is emitted early enough
so that all consumers (display, ringer, etc) have a chance of getting
notified when the call state changes from f.e. DIALING to DISCONNECTED
similar to how its done in 03d960ccaf
for the dummy provider.
2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
17ac56fe44 sip: slightly improved authentication
* removed nua_callstate_authenticating as it was never hit
* move debug statement further up, because we might not reach it if no
  corresponding call is found
* handle 401 and 407 the same way
  note: we should record which realm we're authenticating against during
  REGISTER so we can prompt the user for additional credentials when
  challenged for a different realm - still happens when calling from
  a sip.linphone.org account to a jmp.bwapp.bwsip.io account.
2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
cadaa6a3e0 sip: use g_return_if_fail and friends only for public functions 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
45285062ca sip: improve connection handling by using relevant sofia tags
NUTAG_SUPPORTED and SIPTAG_EXPIRES_STR for instance

sip: include expire, urn uuid

sip.linphone.org accounts seems to be working!
2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
de44a17fe5 sip: use ipv4 exclusively for now
IPv6 should work, but sofia's outbound engine keep printing
errors involving the outbound engine. Working theory:
Failing ICMPv6 (pings) can make sofia think we don't have connectivity.
Note that we also don't get any answers from the SIP servers we tried so far.
2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
dcff7538f2 sip: media: improve SDP offer/answer handling 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
97a925ee39 sip: handle i_outbound 404 errors 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
7b54855f5e sip: media: change default codec to PCMA 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
c33fd53829 sip: Use app name in the user agent 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
1836c7c915 sip: allow specifying local port and use IPv6 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
37b9fe1c30 sip: rework setting SDP 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
afd5b5d6a8 sip: go offline when disposing CallsSipOrigin 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
4521033127 sip: origin: register with SIP server 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
017af5ec8b sip: pipeline: bind sockets for RTP
Add debugging information for used sockets
2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
a53f07dfd3 sip: origin: do not use hardcoded ports for RTP 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
6681077886 sip: origin: emit message on DNS error 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
3133f25c6b sip: call: rework call state changes 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
706a667547 sip: origin: fix direct connection case 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
c9949a5f9f sip: origin: get address on incoming call 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
588f70f78a sip: origin: fix CallsSipHandles reference in sip_callback 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
e0482fc6e6 sip: initial call handling
* implement answering and hangup
 * (de)activate media pipeline
2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
967f30d688 sip: Add media manager and sipify origin
* pipeline: we should bind the used socket of our udpsink to the socket udpsrc
2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
2dfa42d48d sip: sipify provider with sofia 2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
7971fb5afb sip: Origin needs account credentials
Credentials can be set through a config file. The config file is parsed
by CallsSipProvider in order to add origins for each SIP account.
2021-04-03 00:08:31 +02:00
Evangelos Ribeiro Tzaras
71e7a33626 sip: Initial provider
based on dummy provider
2021-04-03 00:08:31 +02:00