SIP Test is flaky
From: https://salsa.debian.org/DebianOnMobile-team/gnome-calls/-/jobs/5300389
==================================== 6/20 ====================================
test: sip
start time: 20:56:39
duration: 0.44s
result: killed by signal 5 SIGTRAP
command: PYTHONDONTWRITEBYTECODE=yes GSETTINGS_SCHEMA_DIR=/builds/DebianOnMobile-team/gnome-calls/debian/output/source_dir/_build/data NO_AT_BRIDGE=1 CALLS_AUDIOSRC=audiotestsrc CALLS_SIP_TEST=1 MALLOC_CHECK_=2 CALLS_AUDIOSINK=fakesink GSETTINGS_BACKEND=memory G_DEBUG=gc-friendly,fatal-warnings CALLS_PLUGIN_DIR=/builds/DebianOnMobile-team/gnome-calls/debian/output/source_dir/_build/plugins MALLOC_PERTURB_=83 ASAN_OPTIONS=halt_on_error=1:abort_on_error=1:print_summary=1 UBSAN_OPTIONS=halt_on_error=1:abort_on_error=1:print_summary=1:print_stacktrace=1 /builds/DebianOnMobile-team/gnome-calls/debian/output/source_dir/_build/plugins/provider/tests/sip
----------------------------------- stdout -----------------------------------
TAP version 13
# random seed: R02S2c3eb21f971ee499c937f65a347132d6
1..5
# Start of Calls tests
# Start of SIP tests
ok 1 /Calls/SIP/provider_object
ok 2 /Calls/SIP/provider_origins
# CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname
# CallsSipOrigin-DEBUG: Account changed:
# user: alice
# host: runner-f0fdd533-project-48009-concurrent-0
# CallsSipMediaManager-DEBUG: Creating CallsSipMediaManager
# GLib-GIO-DEBUG: _g_io_module_get_default: Found default implementation memory (GMemorySettingsBackend) for ?gsettings-backend?
# CallsSettings-DEBUG: Setting country code to
# CallsSettings-DEBUG: Enabling the use of default origins
# CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMA is available
# CallsGstRfc3551-DEBUG: Adding PCMA to the codec candidates
# CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMU is available
# CallsGstRfc3551-DEBUG: Adding PCMU to the codec candidates
# CallsGstRfc3551-DEBUG: Gstreamer plugin for GSM is available
# CallsGstRfc3551-DEBUG: Adding GSM to the codec candidates
# CallsGstRfc3551-DEBUG: Gstreamer plugin for G722 is not available
# CallsGstRfc3551-DEBUG: Gstreamer plugin for G723 is not available
# CallsGstRfc3551-DEBUG: Gstreamer plugin for G722 is not available
# CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMA is available
# CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMU is available
# CallsGstRfc3551-DEBUG: Gstreamer plugin for GSM is available
# CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname
# CallsSipOrigin-DEBUG: Account changed:
# user: bob
# host: runner-f0fdd533-project-48009-concurrent-0
# CallsSipOrigin-DEBUG: Clearing any handles for account 'alice@runner-f0fdd533-project-48009-concurrent-0'
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from NULL to READY
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from NULL to READY
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle
# CallsSipOrigin-DEBUG: Clearing any handles for account 'username@sip.imaginary-host.org'
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle
# CallsSipOrigin-DEBUG: Clearing any handles for account 'bob@runner-f0fdd533-project-48009-concurrent-0'
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle
ok 3 /Calls/SIP/origin_objects
# CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname
# CallsSipOrigin-DEBUG: Account changed:
# user: alice
# host: runner-f0fdd533-project-48009-concurrent-0
# CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname
# CallsSipOrigin-DEBUG: Account changed:
# user: bob
# host: runner-f0fdd533-project-48009-concurrent-0
# CallsSipOrigin-DEBUG: Clearing any handles for account 'bob@runner-f0fdd533-project-48009-concurrent-0'
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle
# CallsSipOrigin-DEBUG: Clearing any handles for account 'alice@runner-f0fdd533-project-48009-concurrent-0'
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle
# CallsSipOrigin-DEBUG: Clearing any handles for account 'username@sip.imaginary-host.org'
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle
ok 4 /Calls/SIP/origin_call_lists
# CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname
# CallsSipOrigin-DEBUG: Account changed:
# user: alice
# host: runner-f0fdd533-project-48009-concurrent-0
# CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname
# CallsSipOrigin-DEBUG: Account changed:
# user: bob
# host: runner-f0fdd533-project-48009-concurrent-0
# DEBUG: Call test: Stage 1
# CallsSipOrigin-DEBUG: Calling `sip:alice@127.0.0.1:5060' from origin 'bob'
# CallsSipOrigin-DEBUG: Setting local SDP for outgoing call to sip:alice@127.0.0.1:5060:
# v=0
# m=audio 52483 RTP/AVP 8 0 3
# a=rtpmap:8 PCMA/8000
# a=rtpmap:0 PCMU/8000
# a=rtpmap:3 GSM/8000
# a=rtcp:54476
#
#
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from NULL to READY
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from NULL to READY
# CallsSipOrigin-DEBUG: incoming call INVITE: 100 Trying from sip:bob@runner-f0fdd533-project-48009-concurrent-0
# CallsSipOrigin-DEBUG: The call state has changed: 100 Trying
# CallsSipOrigin-DEBUG: Found common codec: PCMA
# CallsSipOrigin-DEBUG: Found common codec: PCMU
# CallsSipOrigin-DEBUG: Found common codec: GSM
# CallsSipOrigin-DEBUG: Remote SDP was set to:
# v=0
# o=- 5468622642142409838 1994596778342764480 IN IP4 172.17.0.2
# s=-
# c=IN IP4 172.17.0.2
# t=0 0
# m=audio 52483 RTP/AVP 8 0 3
# a=rtpmap:8 PCMA/8000
# a=rtpmap:0 PCMU/8000
# a=rtpmap:3 GSM/8000
# a=rtcp:54476
#
# CallsSipCall-DEBUG: Setting remote ports: RTP/RTCP 52483/54476
# CallsSipOrigin-DEBUG: Call incoming
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from NULL to READY
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from NULL to READY
# DEBUG: Hanging up call
# CallsSipCall-DEBUG: Hanging up incoming call
# CallsSipOrigin-DEBUG: The call state has changed: 480 Call state
# CallsSipOrigin-DEBUG: Call terminated. Deactivating media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# DEBUG: Call test: Stage 2
# CallsSipOrigin-DEBUG: Calling `sip:bob@127.0.0.1:5061' from origin 'alice'
# CallsSipOrigin-DEBUG: Setting local SDP for outgoing call to sip:bob@127.0.0.1:5061:
# v=0
# m=audio 37173 RTP/AVP 8 0 3
# a=rtpmap:8 PCMA/8000
# a=rtpmap:0 PCMU/8000
# a=rtpmap:3 GSM/8000
# a=rtcp:43906
#
#
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from READY to NULL
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from NULL to READY
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from READY to NULL
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from NULL to READY
# CallsSipOrigin-DEBUG: The call state has changed: 000 INVITE sent
# CallsSipOrigin-DEBUG: The call state has changed: 000 INVITE sent
# CallsSipOrigin-DEBUG: incoming call INVITE: 100 Trying from sip:alice@runner-f0fdd533-project-48009-concurrent-0
# CallsSipOrigin-DEBUG: Cannot handle more than one call. Rejecting
# CallsSipOrigin-DEBUG: The call state has changed: 100 Trying
not ok /Calls/SIP/calls_direct_call - CallsSipOrigin-FATAL-WARNING: No call found for the current handle
Bail out!
----------------------------------- stderr -----------------------------------
su_source_port_create() returns 0x55c26eb59000
su_source_port_create() returns 0x55c26eb59b40
su_source_port_create() returns 0x55c26eb5a1f0
su_source_port_create() returns 0x55c26ec11aa0
su_source_port_create() returns 0x55c26ec11aa0