GOPHONE is single binary CLI SIP Softphone written in GO and powered by diago
gophone dial -media=audio sip:alice@sip.dev.server:5060NOT Open Source, but most of code is on libraries mentioned with exception for some some audio streaming and media setup. This also shows power of this libs. It is mainly developed for simple automated testing purpose, but with Simple API and better integration with other shell tools.
Any feature/bug or supporting/sponzoring is welcomed.
- Linux amd64
- Linux arm64 (no call transcription supported)
- Windows
- macOS amd64 (*NOT SIGNED) Check MacOS malware detect
- macOS arm64 (no call transcription supported) (*NOT SIGNED) Check MacOS malware detec
For audio it expected you have some preinstalled libraries.
- 3 actions: Dial, Answer or Register
- Media IO control: speaker, mic, file
- RTP statistics during call
- Offline Speech To text Transcription of phone call using whisper models
- Provides also json type output for easier filtering and post verification
- OPUS codec support
Roadmap/Features:
- Use flags to automate call scenario of one endpoint
- Dial Answer Register
- Media encoders: ulaw, alaw
- Media IO: rtp logging, speaker, microphone
- RTP stats: packets, packet loss, talking/silence
- Different responses NoAnswer,Busy
- Offline Speech To text Transcription for output.
- Offline Speech To text Transcription for input
- Sending DTMF rfc4733 with delay control
- Simple INTERACTIVE mode Commands via STDIN
- BlindTransfer automation as subcommand of dial/answer
- Opus codec support
- Controling audio formats like pcmu,pcma and opus
- N Step Loadtester integrated with SIP and media metrics
- AttendedTransfer automation as subcommand of dial/answer
- Recording audio to wav for easier visiting later
- Better events logging in Structured logging for easier JSON verification.
- Logging SIP traces to file
- SIP over WebRTC
gophone is single binary so you only need to download and run it.
You can Download from here or here quick links
- Linux amd64 https://github.com/emiago/gophone/releases/latest/download/gophone-linux-amd64
- Linux arm64 https://github.com/emiago/gophone/releases/latest/download/gophone-linux-arm64
- Macos amd64 https://github.com/emiago/gophone/releases/latest/download/gophone-darwin-amd64.tar.gz
- Macos arm64 https://github.com/emiago/gophone/releases/latest/download/gophone-darwin-arm64.tar.gz
Windows builds will be added on request!
wget https://github.com/emiago/gophone/releases/latest/download/gophone-linux-amd64
chmod +x gophone-linux-amd64
mv gophone-linux-amd64 /usr/local/bin/
# There
gophone -hdocker run --rm -it emiago/gophoneDocker version has no Audio and Transcription built in!
Install codec:
# Debian, Ubuntu
sudo apt install libopus0
# Fedora
sudo dnf install opusgophone dial -audio_formats=opus,pcma,pcmu sip:123@sip.server.comIf you need opus binary for your platform please open issue For now it just Linux amd64 exposed
$>gophone -h
gophone is CLI SIP softphone powered by sipgo library.
To find more information about tool and licences visit
https://github.com/emiago/gophone
Usage of gophone command:
Commands:
dial Dial destination.
answer Answer call
register Send register request only without answer.
loadtest N Step call load testing
-t string
Transport udp|tcp|tls|ws|wss (default "udp")
Enviroment variables:
LOG_LEVEL=debug|info|error Log level for output. Default=info
LOG_FORMAT=json|console Log format for output. Default=console
LOG_NOCOLOR Disable color
SIP_DEBUG LOG SIP traffic. Used with LOG_LEVEL=debug
RTP_DEBUG LOG RTP traffic. Used with LOG_LEVEL=debug
RTCP_DEBUG LOG RTCP traffic. Used with LOG_LEVEL=debug
GOPHONE_MEDIA=<same as -media> Sets default media in case of calls
Answer:
gophone answer -l 127.0.0.200:5060
gophone answer -l 127.0.0.200:5060 -code 486 -reason Busy
Answer with register:
gophone answer -ua alice -username alice1234 -password 1234 -register "127.0.0.1:5060"
Dial:
gophone dial sip:1234@127.0.0.200:5060
gophone dial -sipheader="X-AccountId:test123" sip:1234@server:5060
Register:
gophone register -username=sipgo -password=1234 127.0.0.1:5060
With digest authentication:
gophone dial -ua alice -username=alice1234 -password=1234 "sip:echo@server:5060"
With media:
gophone dial -media=audio sip:1234@localhost:5060
gophone dial -media=mic sip:1234@localhost:5060
gophone answer -media=speaker
With transcribe:
gophone dial -media=log -transcribe sip:1234@localhost:5060
With DTMF:
gophone dial -dtmf=79 -dtmf_delay=8s -dtmf_digit_delay=1s -media=speaker sip:1234@localhost:5060
With Recording:
gophone dial -record=test_record.wav sip:1234@localhost:5060
With INTERACTIVE mode:
echo "wait=3s; dtmf=123; hangup;" | gophone dial -i -media=speaker sip:demo@127.0.0.1:5060Running a full call and transcription output at end.
Using json allows some post verification for your call setup.
Useful filtering
gophone ... | jq 'select(.caller=="Dial" and .event=="SIP")'
gophone ... | jq 'select(.caller=="Dial" and .event=="DialogState")'CALLTRANSCRIPTION=$(LOG_FORMAT=json gophone dial -transcribe sip:49123456789@carrier.xy \
| jq -r 'select(.caller=="Transcriber" and .text != null) | .text')
test "Please enter your PIN. Your answer is, 1234." = $CALLTRANSCRIPTIONTo ensure the continued development and maintenance of gophone, this feature is exposed only for supporters(sponsors) of this or other projects in repo. If you are interested having this feature you can contact me on mail and you will receive build
N Step based Load Test for your SIP + RTP media server
Features:
- Providing SIP + RTP metrics during test execution!
- Allows you to define alarm thresholds (ALARM_VALUES) on which call information is dumped, which after you can use to analyze in your PCAPs
- Rate is automatically distributed based on number of steps and rate range
Checkout Demo: https://youtu.be/VXNt7RP4TLc
Load test feature is N Step call based load testing where on each next step your rate of calls is increased. This allows you to see behavior of SIP server under different rate of pressure and call concurrency
Gophone call loadtester.
Media Codec: ulaw, alaw
Transport:UDP,TCP
Usage: loadtest SIPURI
SIPURI format:
exten
sip:user@ip
Flags:
-alarm_rtp_r value
Set alarm threshold as key:value which will dump call when reached. Can be multiple defined or comma separated. ALARM_RTP_VALUES
-alarm_rtp_w value
Set alarm threshold as key:value which will dump call when reached. Can be multiple defined or comma separated. ALARM_RTP_VALUES
-alarm_sip value
Set alarm threshold as key:value which will dump call when reached. Can be multiple defined or comma separated. ALARM_SIP_VALUES
-audio_file string
Play wav audio file. NOTE: sample rate, bit depth must match codec used
-call_duration duration
Duration of call (e.g., 30s, 1m) (default 30s)
-duration duration
Total duration which will be divided by steps (e.g., 30s, 1m) (default 5m0s)
-l string
My listen ip
-rate_interval duration
Rate interval in seconds (default 1s)
-rate_max int
Maximum requests per second (default 10)
-rate_start int
Initial requests per second (default 1)
-rate_steps int
Number of steps to ramp up/down (default 5)
-t string
Transport udp|tcp (default "udp")
-ua string
User agent of phone (default "sipgo")
-ua_auto_inc
Adds suffix number on user agent and auto increments for every new call. Allows to distinct calls in your tests
-ua_host string
User agent host in From header
ALARM_RTP_VALUES:
- pkt_loss:int Default:0 RTP Packet Total Loss
- pkt_loss_fraction_perc:int[0-100] Default:0 RTP Fraction Packet Loss based on RTCP interval
- jitter_ms:int Default:0 RTP Jitter In Milliseconds
ALARM_SIP_VALUES:
- early_hangup:bool Default:false Early hangup by remote side
- max_ring_time:duration Default:0 Max ring time expected
EXAMPLES:
Minimal:
gophone loadtest sip:test@sip.server.xy
Custom rate:
gophone loadtest -rate_start=1 -rate_max=5 -rate_interval=5s sip:test@sip.server.xy
With RTP Alarms:
gophone loadtest -alarm_rtp_r="pkt_loss_perc:5,jitter_ms:25" -alarm_rtp_w="jitter_ms=30" sip:491234566@sip.server.xy
With SIP Alarms:
gophone loadtest -alarm_sip="max_ring_time=5s" sip:491234566@sip.server.xygophone CLI tool is not signed by Apple Developer Program and may be shown as Malware by your Gatekeeper. This is something Apple started to require on newer OS (From Catalina v10.15). FYI: in order to sign binary Apple requires annual payment. This project is not yet sponsored or had requirement todo so.
For allowing unsigned CLI tool to run here are some articles to check.