Hi,
I'm looking to do some audio conferencing using liblinphone but I'm
receiving a seg fault when I start the conference using
linphone_core_add_all_to_conference. The backtrace is below. If I try the
same with linphonec there is also a segmentation fault.
The 3 clients are on liblinphone on linux , linphone on Android and
linphone on Mac. The linux client is crashing and that is where the
conference is initiated.
Any ideas what might be wrong?
Thanks,
Peter
Program terminated with signal SIGSEGV, Segmentation fault.
#0 ms_snd_card_get_minimal_latency (obj=0x0) at base/mssndcard.c:195
195 return obj->latency;
(gdb) backtrace
#0 ms_snd_card_get_minimal_latency (obj=0x0) at base/mssndcard.c:195
#1 0x00007ff83d5d5daa in audio_stream_start_from_io
(stream=stream@entry=0x149cab0,
profile=profile@entry=0x1459db0,
rem_rtp_ip=rem_rtp_ip@entry=0x7ff83dec3c2e "127.0.0.1",
rem_rtp_port=rem_rtp_port@entry=65000,
rem_rtcp_ip=rem_rtcp_ip@entry=0x7ff83dec3c2e
"127.0.0.1",
rem_rtcp_port=rem_rtcp_port@entry=65001, payload=payload@entry=0,
io=io@entry=0x7fffde456070) at voip/audiostream.c:1004
#2 0x00007ff83d5d7178 in audio_stream_start_full (stream=0x149cab0,
profile=0x1459db0, rem_rtp_ip=0x7ff83dec3c2e "127.0.0.1",
rem_rtp_port=65000,
rem_rtcp_ip=0x7ff83dec3c2e "127.0.0.1", rem_rtcp_port=65001, payload=0,
jitt_comp=40, infile=0x0, outfile=0x0, playcard=0x0, captcard=0x0, use_ec=1
'\001')
at voip/audiostream.c:1279
#3 0x00007ff83de6d6a3 in Linphone::LocalConference::addLocalEndpoint
(this=0x1409bb0) at conference.cc:375
#4 0x00007ff83de6d78a in Linphone::LocalConference::addParticipant
(this=0x1409bb0, call=0x13e6570) at conference.cc:408
#5 0x00007ff83de90aab in linphone_core_add_all_to_conference
(lc=0x13b90b0) at linphonecore.c:7879
#6 0x00000000004019cf in call_state_changed (lc=0x13b90b0, call=0x13e6570,
cstate=LinphoneCallStreamsRunning, msg=0x7ff83dec20a7 "Streams running")
at call_conference.c:92
#7 0x00007ff83deaacdb in linphone_core_notify_call_state_changed
(lc=lc@entry=0x13b90b0, call=call@entry=0x13e6570, cstate=cstate@entry
=LinphoneCallStreamsRunning,
message=0x7ff83dec20a7 "Streams running") at vtables.c:83
#8 0x00007ff83de8299d in linphone_call_set_state (call=0x13e6570,
cstate=LinphoneCallStreamsRunning, message=<optimized out>) at
linphonecall.c:1742
#9 0x00007ff83de66cd0 in process_call_accepted (lc=0x13b90b0,
call=0x13e6570, op=<optimized out>) at callbacks.c:563
#10 0x00007ff83deb681f in call_process_response (op_base=0x13c9510,
event=<optimized out>) at bellesip_sal/sal_op_call.c:323
#11 0x00007ff83cf91a06 in belle_sip_client_transaction_notify_response
(t=0x13ed010, resp=<optimized out>) at transaction.c:524
#12 0x00007ff83cf95ae8 in belle_sip_provider_dispatch_response
(msg=0x146c520, p=0x13befa0) at provider.c:214
#13 belle_sip_provider_dispatch_message (prov=0x13befa0, msg=0x146c520) at
provider.c:236
#14 0x00007ff83cf980dd in notify_incoming_messages (obj=0x13cd670) at
channel.c:515
#15 belle_sip_channel_process_stream (obj=0x13cd670, eos=<optimized out>)
at channel.c:621
#16 0x00007ff83cf99e56 in belle_sip_channel_process_read_data
(obj=0x13cd670) at channel.c:659
#17 belle_sip_channel_process_data (obj=obj@entry=0x13cd670,
revents=revents@entry=1) at channel.c:682
#18 0x00007ff83cfa29a8 in on_udp_data (lp=0x13c0510, events=1) at
transports/udp_listeningpoint.c:190
#19 0x00007ff83cf88f80 in belle_sip_main_loop_iterate (ml=0x13bed20) at
belle_sip_loop.c:535
#20 belle_sip_main_loop_run (ml=ml@entry=0x13bed20) at belle_sip_loop.c:590
#21 0x00007ff83cf8923c in belle_sip_main_loop_sleep (ml=0x13bed20,
milliseconds=<optimized out>) at belle_sip_loop.c:602
#22 0x00007ff83cf92fd9 in belle_sip_stack_sleep (stack=<optimized out>,
milliseconds=<optimized out>) at sipstack.c:214
#23 0x00007ff83deb0fe2 in sal_iterate (sal=<optimized out>) at
bellesip_sal/sal_impl.c:817
#24 0x00007ff83de8ed62 in linphone_core_iterate (lc=0x13b90b0) at
linphonecore.c:2800
#25 0x0000000000401df5 in main (argc=4, argv=0x7fffde457978) at
call_conference.c:199/
ubuntu@ie-test-sip:~$ linphonec
2016-11-07 23:40:01:868 ortp-error-Could not find a suitable soundcard !
2016-11-07 23:40:01:868 ortp-error-Could not find a suitable soundcard !
2016-11-07 23:40:01:868 ortp-error-Could not find a suitable soundcard !
Ready
Warning: video is disabled in linphonec, use -V or -C or -D to enable.
linphonec> Refreshing on sip:[email protected]...
linphonec> 2016-11-07 23:40:01:932 ortp-error-belle_sip_get_src_addr_for:
connect() failed: Network is unreachable
2016-11-07 23:40:01:932 ortp-error-Cannot connect to [UDP://
sip.linphone.org:5060]
Registration on sip:[email protected] successful.
linphonec> "Linphone Android" <sip:[email protected]> is
contacting you.
linphonec> Receiving new incoming call from "Linphone Android" <
sip:[email protected]>, assigned id 1
linphonec> answer
Connected.
linphonec> Call 1 with "Linphone Android" <sip:[email protected]>
connected.
2016-11-07 23:40:08:092 ortp-error-Unable to set encryption_mandatory
[0x6b8d98]: srtp support disabled in mediastreamer2
Media streams established with "Linphone Android" <
sip:[email protected]> for call 1 (audio).
linphonec> pause
Pausing call 1 with "Linphone Android" <sip:[email protected]>.
Pausing the current call...
linphonec> linphonec> 2016-11-07 23:40:10:214 ortp-error-Unable to set
encryption_mandatory [0x666de8]: srtp support disabled in mediastreamer2
2016-11-07 23:40:10:215 ortp-error-No dtmf generator at this time !
Call 1 with "Linphone Android" <sip:[email protected]> is now
paused.
linphonec> call sip:[email protected]
Establishing call id to sip:[email protected], assigned id 2
Contacting sip:[email protected]
linphonec> Call 2 to sip:[email protected] in progress.
linphonec> Remote ringing.
linphonec> Remote ringing...
linphonec> Call 2 to sip:[email protected] ringing.
Call 2 with sip:[email protected] connected.
Call answered by sip:[email protected]
linphonec> 2016-11-07 23:40:29:934 ortp-error-Unable to set
encryption_mandatory [0x6ba988]: srtp support disabled in mediastreamer2
Media streams established with sip:[email protected] for call 2
(audio).
linphonec> calls
Call states
Id | Destination | State | Flags |
------------------------------------------------------------------------
1 | "Linphone Android" <sip:[email protected]> | Paused
|
2 | sip:[email protected] | StreamsRunning |
linphonec> conference
Syntax error.
'conference add <call id> : join the call with id 'call id' into the audio
conference.'conference rm <call id> : remove the call with id 'call id'
from the audio conference.
linphonec> conference add 1
Resuming call 1 with "Linphone Android" <sip:[email protected]>.
Resuming the call with "Linphone Android" <sip:[email protected]
>
linphonec> linphonec> Call resumed.
linphonec> 2016-11-07 23:40:40:909 ortp-error-Unable to set
encryption_mandatory [0x666de8]: srtp support disabled in mediastreamer2
Media streams established with "Linphone Android" <
sip:[email protected]> for call 1 (audio).
linphonec> conference add 2
Modifying call parameters...
linphonec> Segmentation fault (core dumped)
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