[Linphone-users] Audio conferencing seg fault

[Peter Sinnott]

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 (address@hidden, address@hidden, 

    address@hidden "127.0.0.1", address@hidden, address@hidden "127.0.0.1", 

    address@hidden, address@hidden, address@hidden) at voip/audiostream.c:1004

#2  0x00007ff83d5d7178 in audio_stream_start_full (stream=0x149cab0, profile="" 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 (address@hidden, address@hidden, address@hidden, 

    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 (address@hidden, address@hidden) 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 (address@hidden) 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/

address@hidden:~$ 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:address@hidden

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:address@hidden successful.

linphonec> "Linphone Android" <sip:address@hidden> is contacting you.

linphonec> Receiving new incoming call from "Linphone Android" <sip:address@hidden>, assigned id 1

linphonec> answer

Connected.

linphonec> Call 1 with "Linphone Android" <sip:address@hidden> 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:address@hidden> for call 1 (audio).

linphonec> pause

Pausing call 1 with "Linphone Android" <sip:address@hidden>.

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:address@hidden> is now paused.

linphonec> call sip:address@hidden

Establishing call id to sip:address@hidden, assigned id 2

Contacting sip:address@hidden

linphonec> Call 2 to sip:address@hidden in progress.

linphonec> Remote ringing.

linphonec> Remote ringing...

linphonec> Call 2 to sip:address@hidden ringing.

Call 2 with sip:address@hidden connected.

Call answered by sip:address@hidden

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:address@hidden for call 2 (audio).

linphonec> calls

Call states

Id |            Destination              |      State      |    Flags   |

------------------------------------------------------------------------

1  | "Linphone Android" <sip:address@hidden> | Paused          | 

2  | sip:address@hidden       | 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:address@hidden>.

Resuming the call with "Linphone Android" <sip:address@hidden>

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:address@hidden> for call 1 (audio).

linphonec> conference add 2

Modifying call parameters...

linphonec> Segmentation fault (core dumped)