[Debian-on-mobile-maintainers] Bug#1019292: gnome-calls FTBFS on IPV6-only buildds

Adrian Bunk bunk at debian.org
Tue Sep 6 23:01:02 BST 2022 

Source: gnome-calls
Version: 0.3.3-1
Severity: serious
Tags: ftbfs

https://buildd.debian.org/status/logs.php?pkg=gnome-calls&arch=amd64
https://buildd.debian.org/status/logs.php?pkg=gnome-calls&arch=all

...
=================================== 14/16 ====================================
test:         sip
start time:   20:46:52
duration:     0.53s
result:       killed by signal 5 SIGTRAP
command:      MALLOC_CHECK_=2 G_TEST_BUILDDIR='/<<PKGBUILDDIR>>/_build/plugins/provider/tests' GSETTINGS_SCHEMA_DIR='/<<PKGBUILDDIR>>/_build/data' GSETTINGS_BACKEND=memory CALLS_AUDIOSRC=audiotestsrc G_TEST_SRCDIR='/<<PKGBUILDDIR>>/plugins/provider/tests' G_DEBUG=gc-friendly,fatal-warnings NO_AT_BRIDGE=1 CALLS_AUDIOSINK=fakesink MALLOC_PERTURB_=243 CALLS_SIP_TEST=1 PYTHONDONTWRITEBYTECODE=yes '/<<PKGBUILDDIR>>/_build/plugins/provider/tests/sip'
----------------------------------- stdout -----------------------------------
# random seed: R02Sad24a1244a453042551428c9a905d7e2
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: x86-conova-01
# 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 G723 is not available
# CallsSipMediaManager-DEBUG: Did not find audio codec G722
# 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: x86-conova-01
# CallsSipOrigin-DEBUG: Clearing any handles
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from NULL to READY
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from NULL to READY
# CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle
# CallsSipOrigin-DEBUG: Clearing any handles
# 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
# 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: x86-conova-01
# CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname
# CallsSipOrigin-DEBUG: Account changed:
# user: bob
# host: x86-conova-01
# CallsSipOrigin-DEBUG: Clearing any handles
# 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
# 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
# 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: x86-conova-01
# CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname
# CallsSipOrigin-DEBUG: Account changed:
# user: bob
# host: x86-conova-01
# DEBUG: Call test: Stage 1
# CallsSipOrigin-DEBUG: Calling `sip:alice at 127.0.0.1:5060' from origin 'bob'
# CallsSipOrigin-DEBUG: Setting local SDP for outgoing call to sip:alice at 127.0.0.1:5060:
# v=0

# m=audio 53611 RTP/AVP 8 0 3

# a=rtpmap:8 PCMA/8000

# a=rtpmap:0 PCMU/8000

# a=rtpmap:3 GSM/8000

# a=rtcp:52537

# 

# 
# 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: The call state has changed: 000 INVITE sent
# CallsSipOrigin-DEBUG: incoming call INVITE: 100 Trying from sip:bob at x86-conova-01
# 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: 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=- 3562168267639768217 5101368152198370217 IN IP6 2a02:16a8:dc41:100::238

# s=-

# c=IN IP6 2a02:16a8:dc41:100::238

# t=0 0

# m=audio 53611 RTP/AVP 8 0 3

# a=rtpmap:8 PCMA/8000

# a=rtpmap:0 PCMU/8000

# a=rtpmap:3 GSM/8000

# a=rtcp:52537

# 
# CallsSipCall-DEBUG: Setting remote ports: RTP/RTCP 53611/52537
# CallsSipOrigin-DEBUG: Call incoming
# 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 at 127.0.0.1:5061' from origin 'alice'
# CallsSipOrigin-DEBUG: Setting local SDP for outgoing call to sip:bob at 127.0.0.1:5061:
# v=0

# m=audio 49100 RTP/AVP 8 0 3

# a=rtpmap:8 PCMA/8000

# a=rtpmap:0 PCMU/8000

# a=rtpmap:3 GSM/8000

# a=rtcp:38085

# 

# 
# CallsSipOrigin-DEBUG: response to outgoing INVITE: 480 Temporarily Unavailable
# CallsSipOrigin-DEBUG: The call state has changed: 480 Temporarily Unavailable
# CallsSipOrigin-DEBUG: Call terminated. Deactivating media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# 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 rtcp-udp-src has changed state from READY to NULL
# 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
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from READY to NULL
# CallsSipOrigin-DEBUG: The call state has changed: 000 INVITE sent
# CallsSipOrigin-DEBUG: incoming call INVITE: 100 Trying from sip:alice at x86-conova-01
# 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=- 9211933157011057919 6832086157911654361 IN IP6 2a02:16a8:dc41:100::238

# s=-

# c=IN IP6 2a02:16a8:dc41:100::238

# t=0 0

# m=audio 49100 RTP/AVP 8 0 3

# a=rtpmap:8 PCMA/8000

# a=rtpmap:0 PCMU/8000

# a=rtpmap:3 GSM/8000

# a=rtcp:38085

# 
# CallsSipCall-DEBUG: Setting remote ports: RTP/RTCP 49100/38085
# 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: Answering incoming call
# CallsSipCall-DEBUG: Setting local SDP to string:
# v=0

# m=audio 57548 RTP/AVP 8 0 3

# a=rtpmap:8 PCMA/8000

# a=rtpmap:0 PCMU/8000

# a=rtpmap:3 GSM/8000

# a=rtcp:41973

# 

# 
# DEBUG: Hanging up call
# CallsSipCall-DEBUG: Hanging up ongoing call
# CallsSipOrigin-DEBUG: The call state has changed: 200 Call state
# CallsSipOrigin-DEBUG: response to outgoing INVITE: 200 OK
# CallsSipOrigin-DEBUG: The call state has changed: 200 OK
# CallsSipOrigin-DEBUG: Found common codec: PCMA
# CallsSipOrigin-DEBUG: Remote SDP was set to:
# v=0

# o=- 1371930777682465956 1749628799533106578 IN IP6 2a02:16a8:dc41:100::238

# s=-

# c=IN IP6 2a02:16a8:dc41:100::238

# t=0 0

# m=audio 57548 RTP/AVP 8

# a=rtpmap:8 PCMA/8000

# a=rtcp:41973

# 
# CallsSipCall-DEBUG: Setting remote ports: RTP/RTCP 57548/41973
# CallsSipOrigin-DEBUG: Call ready. Activating media pipeline
# CallsSipCall-DEBUG: Setting codec 'PCMA' for pipeline
# CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMA is available
# CallsSipMediaPipeline-DEBUG: Capabilities:
# application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA,payload=(int)8
# CallsSipMediaPipeline-DEBUG: Starting media pipeline
# CallsSipMediaPipeline-DEBUG: RTP/RTCP port before starting pipeline: 49100/38085
# CallsSipMediaPipeline-DEBUG: RTP/RTCP port after starting pipeline: 49100/38085
# CallsSipOrigin-DEBUG: incoming ACK: 200 OK
# CallsSipOrigin-DEBUG: The call state has changed: 200 OK
# CallsSipOrigin-DEBUG: Call ready. Activating media pipeline
# CallsSipCall-DEBUG: Setting codec 'PCMA' for pipeline
# CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMA is available
# CallsSipMediaPipeline-DEBUG: Capabilities:
# application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA,payload=(int)8
# CallsSipMediaPipeline-DEBUG: Starting media pipeline
# CallsSipMediaPipeline-DEBUG: RTP/RTCP port before starting pipeline: 57548/41973
# CallsSipMediaPipeline-DEBUG: RTP/RTCP port after starting pipeline: 57548/41973
# CallsSipOrigin-DEBUG: response to BYE: 200 OK
# CallsSipOrigin-DEBUG: The call state has changed: 200 to BYE
# CallsSipOrigin-DEBUG: Call terminated. Deactivating media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# CallsSipOrigin-DEBUG: incoming BYE: 200 Session Terminated
# CallsSipOrigin-DEBUG: The call state has changed: 200 Session Terminated
# CallsSipOrigin-DEBUG: Call terminated. Deactivating media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
Bail out! CallsSipMediaPipeline-FATAL-WARNING: Error on the message bus: Could not get/set settings from/on resource. (../gst/udp/gstmultiudpsink.c(1228): gst_multiudpsink_configure_client (): /GstPipeline:media-pipeline/GstUDPSink:rtcp-udp-sink:
Invalid address family (got 10))
----------------------------------- stderr -----------------------------------
su_source_port_create() returns 0x558fc04f60c0
su_source_port_create() returns 0x558fc04eecc0
su_source_port_create() returns 0x558fc04f60c0
sres: /etc/resolv.conf: unknown option         
sres: /etc/resolv.conf: unknown option         
sres: /etc/resolv.conf: unknown option         
su_source_port_create() returns 0x558fc04f60c0
sres: /etc/resolv.conf: unknown option         
sres: /etc/resolv.conf: unknown option         
sres: /etc/resolv.conf: unknown option         
su_source_port_create() returns 0x558fc04f60c0
sres: /etc/resolv.conf: unknown option         
sres: /etc/resolv.conf: unknown option         
sres: /etc/resolv.conf: unknown option         

(/<<PKGBUILDDIR>>/_build/plugins/provider/tests/sip:3109934): CallsSipMediaPipeline-WARNING **: 20:46:52.860: Error on the message bus: Could not get/set settings from/on resource. (../gst/udp/gstmultiudpsink.c(1228): gst_multiudpsink_configure_client (): /GstPipeline:media-pipeline/GstUDPSink:rtcp-udp-sink:
Invalid address family (got 10))
==============================================================================
...
Summary of Failures:

14/16 sip                          FAIL            0.53s   killed by signal 5 SIGTRAP

Ok:                 15  
Expected Fail:      0   
Fail:               1   
Unexpected Pass:    0   
Skipped:            0   
Timeout:            0   
dh_auto_test: error: cd _build && LC_ALL=C.UTF-8 MESON_TESTTHREADS=4 meson test returned exit code 1
make[1]: *** [debian/rules:29: override_dh_auto_test] Error 25

More information about the Debian-on-mobile-maintainers mailing list