(See the README.md file in the upper level 'examples' directory for more information about examples.)
This example allows users to make calls over the Internet.
This example is will run on boards marked with green checkbox. Please remember to select the board in menuconfig as discussed is section Usage below.
| Board Name | Getting Started | Chip | Compatible |
|---|---|---|---|
| ESP32-LyraT |  |
 |
 |
| ESP32-LyraTD-MSC |  |
|
|
| ESP32-LyraT-Mini |  |
|
|
| ESP32-Korvo-DU1906 |  |
|
 |
| ESP32-S2-Kaluga-1 Kit |  |
 |
|
Open the project configuration menu (make menuconfig or idf.py menuconfig).
- Select compatible audio board in
Audio HAL. - Set up Wi-Fi connection in
VoIP App Configuration>WiFi SSIDandWiFi Password. - Select compatible audio codec in
VoIP App Configuration>SIP Codec. - Set up SIP URI (Transport://user:password@server:port) in
VoIP App Configuration>SIP_URI.
You can choose one of these servers to build, but we recommend FreePBX.
- Asterisk FreePBX
- Asterisk for Raspberry Pi
- Freeswitch
- Kamailio
- Yate Server
Build the project and flash it to the board, then run monitor tool to view serial output:
make -j8 flash monitor
Or, for CMake based build system (replace PORT with serial port name):
idf.py -p PORT flash monitor
(To exit the serial monitor, type Ctrl-].)
See the Getting Started Guide for full steps to configure and use ESP-IDF to build projects.
- Lightweight
- Support multiple transports for SIP (UDP, TCP, TLS)
- Support G711A/8000 & G711U/8000 Audio Codec
- Easy setting up by using URI
- Tested PBX Server list :
- Asterisk FreePBX v14 / v15
- Freeswitch v1.8.5
- Yate v6.0
- Kamailio v5.3
I (4164) VOIP_EXAMPLE: [ 3 ] Create and start input key service
I (4164) VOIP_EXAMPLE: [ 4 ] Create SIP Service
I (4184) SIP: Conecting...
W (4184) SIP: CHANGE STATE FROM 0, TO 1, :func: sip_connect:1564
I (4184) SIP: [1970-01-01/00:00:01]=======WRITE 0587 bytes>>
I (4184) SIP:
REGISTER sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.51:16810;branch=z9hG4bK--1374356523;rport
From: <sip:[email protected]:5060>;tag=319434100
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:16810>
Max-Forwards: 70
Call-ID: 92FA07B61E9DFFAEA89F08C3F382F7AF342CE46DC71F
CSeq: 1 REGISTER
Expires: 3600
User-Agent: ESP32 SIP/2.0
Content-Length: 0
Allow: INVITE, ACK, CANCEL, BYE, UPDATE, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
Allow-Events: presence, kpml
I (4244) SIP: [1970-01-01/00:00:01]=======================>>
I (4294) SIP: [1970-01-01/00:00:01]<<=====READ 0612 bytes==
I (4294) SIP:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.50.51:16810;branch=z9hG4bK--1374356523;rport=16810
From: <sip:[email protected]:5060>;tag=319434100
To: <sip:[email protected]:5060>;tag=DFvg6Q9830p9a
Call-ID: 92FA07B61E9DFFAEA89F08C3F382F7AF342CE46DC71F
CSeq: 1 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.8.5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
WWW-Authenticate: Digest realm="192.168.50.50", nonce="f5d78ba3-36fd-4726-a4bb-a669fa0c575c", algorithm=MD5, qop="auth"
Content-Length: 0
I (4344) SIP: [1970-01-01/00:00:01]<<======================
I (4354) SIP: Required authentication
I (4354) SIP: [1970-01-01/00:00:01]=======WRITE 0837 bytes>>
I (4364) SIP:
REGISTER sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.51:16810;branch=z9hG4bK-260131642;rport
From: <sip:[email protected]:5060>;tag=-1948762171
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:16810>
Max-Forwards: 70
Call-ID: 92FA07B61E9DFFAEA89F08C3F382F7AF342CE46DC71F
CSeq: 2 REGISTER
Expires: 3600
User-Agent: ESP32 SIP/2.0
Content-Length: 0
Allow: INVITE, ACK, CANCEL, BYE, UPDATE, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
Allow-Events: presence, kpml
Authorization: Digest username="1000", realm="192.168.50.50", nonce="f5d78ba3-36fd-4726-a4bb-a669fa0c575c", uri="sip:192.168.50.50:5060", response="3394d8e71a7d36abcced2a0912e3ff5c", algorithm=MD5, nc=00000001, cnonce="6fe907854a9bc351", qop="auth"
I (4434) SIP: [1970-01-01/00:00:01]=======================>>
I (4464) SIP: [1970-01-01/00:00:01]<<=====READ 0572 bytes==
I (4464) SIP:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.51:16810;branch=z9hG4bK-260131642;rport=16810
From: <sip:[email protected]:5060>;tag=-1948762171
To: <sip:[email protected]:5060>;tag=erN97jtc19cvp
Call-ID: 92FA07B61E9DFFAEA89F08C3F382F7AF342CE46DC71F
CSeq: 2 REGISTER
Contact: <sip:[email protected]:16810>;expires=3600
Date: Tue, 16 Jun 2020 12:09:04 GMT
User-Agent: FreeSWITCH-mod_sofia/1.8.5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Length: 0
I (4514) SIP: [1970-01-01/00:00:01]<<======================
I (4524) VOIP_EXAMPLE: SIP_EVENT_REGISTERED
W (4524) SIP: CHANGE STATE FROM 1, TO 2, :func: sip_register:1592
I (688034) SIP: [1970-01-01/00:11:13]<<=====READ 1372 bytes==
I (688034) SIP:
INVITE sip:[email protected]:16810 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.50;rport;branch=z9hG4bKBXFpg22KXpmHj
Max-Forwards: 69
From: "Extension 1001" <sip:[email protected]>;tag=9B06QD1BKt6mF
To: <sip:[email protected]:16810>
Call-ID: 9a8db3e8-2a6e-1239-0d93-0b85a3e82ded
CSeq: 21601221 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: FreeSWITCH-mod_sofia/1.8.5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 443
X-FS-Support: update_display,send_info
Remote-Party-ID: "Extension 1001" <sip:[email protected]>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1592283967 1592283968 IN IP4 192.168.50.50
s=FreeSWITCH
c=IN IP4 192.168.50.50
t=0 0
m=audio 26060 RTP/AVP 8 0 96 97 98 101 102 104
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 SPEEX/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:98 SPEEX/32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/16000
a=fmtp:102 0-16
a=rtpmap:104 telephone-event/32000
a=fmtp:104 0-16
a=ptime:20
I (688154) SIP: [1970-01-01/00:11:13]<<======================
I (688164) SIP: Remote RTP port=26060
I (688164) SIP: Remote RTP addr=192.168.50.50
I (688164) SIP: call from 1001
I (688174) SIP: [1970-01-01/00:11:13]=======WRITE 0414 bytes>>
I (688174) SIP:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.50.50;rport;branch=z9hG4bKBXFpg22KXpmHj
Contact: <sip:[email protected]:16810>
From: "Extension 1001" <sip:[email protected]>;tag=9B06QD1BKt6mF
To: <sip:[email protected]:16810>;tag=645213922
Call-ID: 9a8db3e8-2a6e-1239-0d93-0b85a3e82ded
CSeq: 21601221 INVITE
Server: ESP32 SIP/2.0
Allow: ACK, INVITE, BYE, UPDATE, CANCEL, OPTIONS, INFO
Content-Length: 0
I (688214) SIP: [1970-01-01/00:11:13]=======================>>
I (688224) SIP: [1970-01-01/00:11:13]=======WRITE 0417 bytes>>
I (688234) SIP:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.50.50;rport;branch=z9hG4bKBXFpg22KXpmHj
Contact: <sip:[email protected]:16810>
From: "Extension 1001" <sip:[email protected]>;tag=9B06QD1BKt6mF
To: <sip:[email protected]:16810>;tag=-1799498926
Call-ID: 9a8db3e8-2a6e-1239-0d93-0b85a3e82ded
CSeq: 21601221 INVITE
Server: ESP32 SIP/2.0
Allow: ACK, INVITE, BYE, UPDATE, CANCEL, OPTIONS, INFO
Content-Length: 0
I (688274) SIP: [1970-01-01/00:11:13]=======================>>
W (688284) SIP: CHANGE STATE FROM 2, TO 16, :func: _sip_uas_process_req_invite:810
I (689384) VOIP_EXAMPLE: ringing... RemotePhoneNum 1001
I (690384) VOIP_EXAMPLE: ringing... RemotePhoneNum 1001
I (691384) VOIP_EXAMPLE: ringing... RemotePhoneNum 1001
I (692384) VOIP_EXAMPLE: ringing... RemotePhoneNum 1001
I (1372804) SIP: User call sip BYE
I (1372834) SIP: [1970-01-01/00:22:25]=======WRITE 0664 bytes>>
I (1372834) SIP:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.50;rport;branch=z9hG4bKDF27Kr4tQ80pS
From: <sip:[email protected]:16810>;tag=1647388708
To: "Extension 1001" <sip:[email protected]>;tag=6HXNBy9HZvcKF
Contact: <sip:[email protected]:16810>
Max-Forwards: 70
Call-ID: 2f65a33b-2a70-1239-0d93-0b85a3e82ded
CSeq: 43 BYE
Expires: 3600
User-Agent: ESP32 SIP/2.0
Content-Length: 0
Authorization: Digest username="1000", realm="192.168.50.50", nonce="f5d78ba3-36fd-4726-a4bb-a669fa0c575c", uri="sip:192.168.50.50:5060", response="f8c3de6f561f54dec9498a28f2fb5c73", algorithm=MD5, nc=0000002a, cnonce="19f9e2e22ddc905c", qop="auth"
I (1372884) SIP: [1970-01-01/00:22:25]=======================>>
I (1372914) SIP: [1970-01-01/00:22:25]<<=====READ 0501 bytes==
I (1372914) SIP:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.50;rport=16810;branch=z9hG4bKDF27Kr4tQ80pS;received=192.168.50.51
From: <sip:[email protected]:16810>;tag=1647388708
To: "Extension 1001" <sip:[email protected]>;tag=6HXNBy9HZvcKF
Call-ID: 2f65a33b-2a70-1239-0d93-0b85a3e82ded
CSeq: 43 BYE
User-Agent: FreeSWITCH-mod_sofia/1.8.5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Length: 0
I (1372954) SIP: [1970-01-01/00:22:25]<<======================
W (1372964) SIP: CHANGE STATE FROM 32, TO 2, :func: _sip_request_bye:1368
I (1372974) SIP_RTP: send task stopped
I (1373024) VOIP_EXAMPLE: SIP_EVENT_AUDIO_SESSION_END
- If you need reduce the RTT(Response time), you can set esp_log_level_set("SIP", ESP_LOG_WARN);