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MayamaTakeshi
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sip-lab
A node module that helps to write SIP functional tests
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Enable codec opus
#121
MayamaTakeshi
closed
2 weeks ago
1
Add proper error message indicating no timeout was passed to z.wait() and z.sleep()
#120
MayamaTakeshi
opened
1 month ago
0
Reimplement BFSK support using spandsp
#119
MayamaTakeshi
opened
2 months ago
0
Review media function and codec selection
#118
MayamaTakeshi
opened
2 months ago
0
Properly implement sip.stop() to fully stop the underlying pjsip/pjmedia engines
#117
MayamaTakeshi
opened
2 months ago
0
sip.stop(cleanup=true) should check for calls, registrations, subscription counters to become zero
#116
MayamaTakeshi
opened
2 months ago
0
Permit to send UPDATE with media.
#115
MayamaTakeshi
opened
2 months ago
0
Permit to reply to UPDATE requests
#114
MayamaTakeshi
opened
2 months ago
0
Permit to set specific from_tag when creating call
#113
MayamaTakeshi
opened
2 months ago
0
Implement new function call.update
#112
MayamaTakeshi
closed
2 months ago
1
Permit to set specific Call-ID when creating call
#111
MayamaTakeshi
closed
2 months ago
1
Try to remove Supported: 100rel
#110
MayamaTakeshi
opened
2 months ago
0
Remove default header Supported
#109
MayamaTakeshi
closed
2 months ago
3
Add support for BFSK generation/detection
#108
MayamaTakeshi
closed
2 months ago
1
Fails to build when using node v21.7.3
#107
MayamaTakeshi
opened
5 months ago
1
addon build fails on debian 10 (buster)
#106
MayamaTakeshi
opened
5 months ago
0
samples/tls.js segfaults on docker container and virtual machine
#105
MayamaTakeshi
opened
5 months ago
3
allocate_tls_tpfactory allowing creation transport using the same port
#104
MayamaTakeshi
opened
5 months ago
1
Occasionally options.js fails with 'Cannot respond to our own request'
#103
MayamaTakeshi
opened
5 months ago
1
Check if dtmf detection works with samplingRate different of 8000
#102
MayamaTakeshi
opened
6 months ago
0
Cannot use more than one websocket client
#101
MayamaTakeshi
closed
5 months ago
0
Occasional crash when running 'node samples/pcma.js'
#100
MayamaTakeshi
closed
6 months ago
1
Occasional crash when running 'node samples/g729.js'
#99
MayamaTakeshi
closed
6 months ago
2
send_dtmf (for inband at least) should permit to specify digit duration and interdigit duration
#98
MayamaTakeshi
opened
8 months ago
0
Failure to resolve transport for incoming call when names are used in case of ip addresses
#97
MayamaTakeshi
closed
8 months ago
1
Correct processing of call_ended
#96
MayamaTakeshi
closed
8 months ago
1
Add support for ipv6
#95
MayamaTakeshi
opened
8 months ago
0
Replace addon_log with PJ_PERROR
#94
MayamaTakeshi
opened
8 months ago
0
Prevent restart of audio media stream if media didn't change
#93
MayamaTakeshi
opened
8 months ago
0
Regarding recreation of pjmedia_ports when reinvite causes change in stream port characteristics
#91
MayamaTakeshi
opened
8 months ago
0
Codec ilbc not working
#90
MayamaTakeshi
opened
8 months ago
1
Refactor stop media functions
#89
MayamaTakeshi
closed
8 months ago
0
Correct implementation of parameter media_id in media functions
#88
MayamaTakeshi
closed
8 months ago
1
wav recording doesn't survive a reinvite
#87
MayamaTakeshi
closed
8 months ago
1
start_speech_synth should permit to specify option end_of_speech_event
#85
MayamaTakeshi
closed
8 months ago
0
start_play_wav should permit to specify option end_of_file_event
#84
MayamaTakeshi
closed
8 months ago
0
test samples/sip_cancel.js occasional crash
#83
MayamaTakeshi
closed
8 months ago
3
Convert sip-lab to a generic sip library
#82
MayamaTakeshi
opened
8 months ago
0
Add support for webrtc
#81
MayamaTakeshi
opened
8 months ago
2
Add support for websocket server to start_speech_recog
#80
MayamaTakeshi
closed
5 months ago
0
Add support for websocket server to start_speech_synth
#79
MayamaTakeshi
closed
6 months ago
1
Implement stop_speech_synth
#78
MayamaTakeshi
closed
8 months ago
0
Implement notification of end_of_speech in flite_port (speech_synth)
#77
MayamaTakeshi
closed
8 months ago
0
Implement support for loop in start_speech_synth
#76
MayamaTakeshi
closed
8 months ago
0
Create conf per AudioEndpoint
#74
MayamaTakeshi
closed
8 months ago
1
Samples should not reference hardcoded port numbers (SIP and RTP).
#73
MayamaTakeshi
opened
9 months ago
1
Solve warnings
#72
MayamaTakeshi
closed
8 months ago
0
set_codecs only setting a single codec
#70
MayamaTakeshi
closed
9 months ago
0
npm i sip-lab fails if build dependencies are not installed
#69
MayamaTakeshi
opened
9 months ago
0
Failing with node versions older than 19
#68
MayamaTakeshi
closed
9 months ago
2
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