How to play audio file? about gosip

[jart]

There's a command line tool called sox that makes it easy to convert audio files to the format needed by gosip, which is signed linear 16-bit samples at 8khz. For example:

wget https://justine.storage.googleapis.com/numbers.mp3
sox numbers.mp3 -c1 -traw -esigned -b16 -r8000 numbers.s16
play numbers.s16

To get started playing audio files in a telephone calls, you can modify the fone/main.go example so it reads audio from a file rather than your microphone. The ticker.C channel is triggered every 20 milliseconds. Each time that happens, you need to read 320 bytes from the file and pass it along to rtp.Send().

I recommend the audio files be converted to s16 beforehand, so decoding is a one-time cost. If you want to play arbitrary audio formats on the fly, one thing that works great is spawning sox as a subprocess and reading the samples via a pipe. That should minimize the chance of cpu-bound audio decoding disrupting the event loop, which for PSTN calls must behave like real time.

[yaosiqi525]

Can you provide the param format?
or For example, like "username@host"
main.go need "requestURIString" param

[jart]

The URI format depends.

If you're calling the PSTN you need to use a service like Flowroute. If you do that they'll give you a prefix. If you wanted to dial a number in New York it'd look something like this:

$ fone sip:12345678*[email protected]

If you want to dial a software telephone, usually they'll speak sip natively and the SIP URI will end up looking more like an email address.

[yaosiqi525]

me try "go run ./fone/main.go sip:[email protected]"
but it return:
panic: Connection refused
when run "makePulseAudio(C.PA_STREAM_RECORD, requestURIString)"

me running in docker,whether the influence?

[jart]

I would recommend contacting your local administrator and troubleshooting the problem on your end.

[yaosiqi525]

me find fone/main.go, but only find use mic
how to read music file? me generated the numbers.s16, but i don't know the way to open it

[jart]

Open it as a normal file using the Go standard library.

Please note this project supports gosip but we can't provide technical support in general.

[yaosiqi525]

Thank you very much for your answer

[hoangtuan151]

Thx you @jart for your awesome library!

I'm using the echo_test example code to make a SIP phone call from a Go application to another soft phone. I have FreeSwitch configured, and the Go app successfully send INVITE to soft phone, but I have trouble and really need your help: I can't hear anything from soft phone

Some clues

1/ Here is my modified code
Note: I try so set size=320 (at line 159) but it's not work!

2/ The audio file is in right format:

play ../assets/bot_greeting.wav         
play WARN alsa: can't encode 0-bit Unknown or not applicable

../assets/bot_greeting.wav:

 File Size: 72.2k     Bit Rate: 128k
  Encoding: Signed PCM    
  Channels: 1 @ 16-bit   
Samplerate: 8000Hz       
Replaygain: off         
  Duration: 00:00:04.51  

In:100%  00:00:04.51 [00:00:00.00] Out:36.1k [      |      ] Hd:5.1 Clip:0    
Done.

Note: I also try these but no luck:

• Using numbers.s16 audio file
• Ulaw converting: frameout[rtp.HeaderSize+n] = byte(dsp.LinearToUlaw(int64(raw_audio[raw_audio_idx])))

3/ Application log

➜  test git:(dev) ✗ go run .
test:main() begin...
2022/11/15 17:59:55 >>> 10.124.68.213:5080
INVITE sip:[email protected]:5080 SIP/2.0
From: Echo Test <sip:10.124.68.213:33684>;tag=ab3072e56f7f
To: <sip:10.124.68.213:5080>
Via: SIP/2.0/UDP 10.124.68.213:33684;branch=z9hG4bK-e83efd2990b5
Contact: <sip:10.124.68.213:33684>
Call-ID: 912685b5-8ce3-469e-a0ce-4537ebe4d35c
CSeq: 17257 INVITE
User-Agent: gosip/1.o
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 240

v=0
o=- 1929021139 1929021139 IN IP4 10.124.68.213
s=my people call themselves dark angels
c=IN IP4 10.124.68.213
t=0 0
m=audio 49184 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

2022/11/15 17:59:55 <<< 10.124.68.213:5080
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.124.68.213:33684;branch=z9hG4bK-e83efd2990b5
From: Echo Test <sip:10.124.68.213:33684>;tag=ab3072e56f7f
To: <sip:10.124.68.213:5080>
Call-ID: 912685b5-8ce3-469e-a0ce-4537ebe4d35c
CSeq: 17257 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git~20200818T185121Z~25569c1631~64bit
Content-Length: 0


2022/11/15 17:59:58 <<< 10.124.68.213:5080
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.124.68.213:33684;branch=z9hG4bK-e83efd2990b5
From: Echo Test <sip:10.124.68.213:33684>;tag=ab3072e56f7f
To: <sip:10.124.68.213:5080>;tag=Q0UUBv4N3BSjQ
Call-ID: 912685b5-8ce3-469e-a0ce-4537ebe4d35c
CSeq: 17257 INVITE
Contact: <sip:[email protected]:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git~20200818T185121Z~25569c1631~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 222
Remote-Party-ID: "1001" <sip:[email protected]>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1668493614 1668493615 IN IP4 10.124.68.213
s=FreeSWITCH
c=IN IP4 10.124.68.213
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2022/11/15 17:59:58 wanted 200 ok but got:183Session Progress... retry in 5s
2022/11/15 18:00:03 <<< 10.124.68.213:5080
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.124.68.213:33684;branch=z9hG4bK-e83efd2990b5
From: Echo Test <sip:10.124.68.213:33684>;tag=ab3072e56f7f
To: <sip:10.124.68.213:5080>;tag=Q0UUBv4N3BSjQ
Call-ID: 912685b5-8ce3-469e-a0ce-4537ebe4d35c
CSeq: 17257 INVITE
Contact: <sip:[email protected]:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git~20200818T185121Z~25569c1631~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 222
Remote-Party-ID: "Outbound Call" <sip:[email protected]>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1668493614 1668493615 IN IP4 10.124.68.213
s=FreeSWITCH
c=IN IP4 10.124.68.213
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2022/11/15 18:00:03 read 72238 bytes from sample audio
2022/11/15 18:00:04 written at audio idx: 8000
2022/11/15 18:00:05 written at audio idx: 16000
2022/11/15 18:00:06 written at audio idx: 24000
2022/11/15 18:00:07 written at audio idx: 32000
2022/11/15 18:00:08 written at audio idx: 40000
2022/11/15 18:00:09 written at audio idx: 48000
2022/11/15 18:00:10 written at audio idx: 56000
2022/11/15 18:00:11 written at audio idx: 64000
2022/11/15 18:00:12 written at audio idx: 72000
2022/11/15 18:00:12 Going to hangup...
2022/11/15 18:00:12 End call !!!
test:main() end!

[hoangtuan151]

Hi @yaosiqi525 , did you get it work with your audio file?

[tong3jie]

@yaosiqi525 @hoangtuan151 you can try to get this library(zaf/g711)