[Debian-on-mobile-maintainers] Bug#1017068: gnome-calls: sip test fails when builds are running on machines without network
Shannon Brady
shannonbrady at google.com
Fri Aug 12 21:30:09 BST 2022
Package: gnome-calls
X-Debbugs-Cc: shannonbrady at google.com
Version: 43~alpha.2-1
Severity: important
Dear Maintainer,
When building this version of gnome-call on a system that is not connected
to the network, builds will always fail the SIP test with the following
error
15/16 sip FAIL 0.40s killed by signal
5 SIGTRAP
12:24:34 G_TEST_BUILDDIR='/<<PKGBUILDDIR>>/_build/tests'
G_DEBUG=gc-friendly,fatal-warnings PYTHONDONTWRITEBYTECODE=yes
G_TEST_SRCDIR='/<<PKGBUILDDIR>>/tests'
GSETTINGS_SCHEMA_DIR='/<<PKGBUILDDIR>>/_build/data' NO_AT_BRIDGE=1
CALLS_AUDIOSINK=fakesink MALLOC_CHECK_=2 MALLOC_PERTURB_=140
CALLS_AUDIOSRC=audiotestsrc CALLS_SIP_TEST=1 GSETTINGS_BACKEND=memory
'/<<PKGBUILDDIR>>/_build/tests/sip'
----------------------------------- output
-----------------------------------
stdout:
# random seed: R02S46e8f6042a6f29398f786aae28ab4bfb
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: ubuntu
# 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: ubuntu
# 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: ubuntu
# CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname
# CallsSipOrigin-DEBUG: Account changed:
# user: bob
# host: ubuntu
# 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: ubuntu
# CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname
# CallsSipOrigin-DEBUG: Account changed:
# user: bob
# host: ubuntu
# 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 54401 RTP/AVP 8 0 3
# a=rtpmap:8 PCMA/8000
# a=rtpmap:0 PCMU/8000
# a=rtpmap:3 GSM/8000
# a=rtcp:58045
#
#
# 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 ubuntu
# 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=- 3944166252790087204 7751455665547847975 IN IP6 fd00::1
# s=-
# c=IN IP6 fd00::1
# t=0 0
# m=audio 54401 RTP/AVP 8 0 3
# a=rtpmap:8 PCMA/8000
# a=rtpmap:0 PCMU/8000
# a=rtpmap:3 GSM/8000
# a=rtcp:58045
#
# CallsSipCall-DEBUG: Setting remote ports: RTP/RTCP 54401/58045
# 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 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 33023 RTP/AVP 8 0 3
# a=rtpmap:8 PCMA/8000
# a=rtpmap:0 PCMU/8000
# a=rtpmap:3 GSM/8000
# a=rtcp:48415
#
#
# 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
# CallsSipOrigin-DEBUG: The call state has changed: 000 INVITE sent
# 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: incoming call INVITE: 100 Trying from
sip:alice at ubuntu
# 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=- 1039427969357689109 1694529409539312076 IN IP6 fd00::1
# s=-
# c=IN IP6 fd00::1
# t=0 0
# m=audio 33023 RTP/AVP 8 0 3
# a=rtpmap:8 PCMA/8000
# a=rtpmap:0 PCMU/8000
# a=rtpmap:3 GSM/8000
# a=rtcp:48415
#
# CallsSipCall-DEBUG: Setting remote ports: RTP/RTCP 33023/48415
# 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 59445 RTP/AVP 8 0 3
# a=rtpmap:8 PCMA/8000
# a=rtpmap:0 PCMU/8000
# a=rtpmap:3 GSM/8000
# a=rtcp:35082
#
#
# 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=- 4515355432483596688 3044272662504464665 IN IP6 fd00::1
# s=-
# c=IN IP6 fd00::1
# t=0 0
# m=audio 59445 RTP/AVP 8
# a=rtpmap:8 PCMA/8000
# a=rtcp:35082
#
# CallsSipCall-DEBUG: Setting remote ports: RTP/RTCP 59445/35082
# 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:
33023/48415
# CallsSipMediaPipeline-DEBUG: RTP/RTCP port after starting pipeline:
33023/48415
# 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
# 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:
59445/35082
# CallsSipMediaPipeline-DEBUG: RTP/RTCP port after starting pipeline:
59445/35082
# 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
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 0x572b8d34
su_source_port_create() returns 0x572bb434
su_source_port_create() returns 0x572c7c34
sres: /etc/resolv.conf: unknown option
sres: /etc/resolv.conf: unknown option
sres: /etc/resolv.conf: unknown option
su_source_port_create() returns 0x572c0134
sres: /etc/resolv.conf: unknown option
sres: /etc/resolv.conf: unknown option
sres: /etc/resolv.conf: unknown option
su_source_port_create() returns 0x572c7c34
sres: /etc/resolv.conf: unknown option
sres: /etc/resolv.conf: unknown option
sres: /etc/resolv.conf: unknown option
(/<<PKGBUILDDIR>>/_build/tests/sip:50791): CallsSipMediaPipeline-WARNING
**: 12:24:34.431: 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))
------------------------------------------------------------------------------
Note that address_family 10 corresponds to AF_INT6, which is odd because
the addresses that are assigned in the test are IPv4 addresses (
https://gitlab.gnome.org/GNOME/calls/-/blob/master/tests/test-sip.c#L274) .
Specifically 127.0.0 which is IPv4 localhost
However, this test will succeed on a build network that does have network
and DNS.Is this expected behaviour? I would have assumed that builds should
be able to succeed without internet connection.
Thanks,
Shannon
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