Skip to content

Latest commit

 

History

History
34 lines (30 loc) · 1.97 KB

testing.md

File metadata and controls

34 lines (30 loc) · 1.97 KB
Raw

Testing the SIPMediaGW

Once the services are up and running, you can join a conference from your preferred SIP softphone by calling the following SIP addresses :

Testing with Baresip

We provide a bash script that runs a SIP call via Baresip. To launch the script, simply replace KAMAILIO_IP_ADDRESS with the corresponding IP address and run the following command:

./test/baresip/SIPCall.sh -u sip:test@YOUR_IP -d 0@KAMAILIO_IP_ADDRESS

Testing with Linphone

It is also possible to perform tests from a SIP client such as Linphone.

  1. Download and install Linphone
  2. Add an account (in Preferences) [1] [2] [3] [4]
    1. SIP Address: sip:username@YOUR_IP
    2. SIP Server address : <sip:YOUR_IP;transport=tls>
    3. Disable the following options : Register, Publish presence information, Enable AVPF, Enable ICE, Bundle mode
  3. In the audio section, it is recommended to disable all codecs except PCMU, PCMA and G722
  4. In the video section, it is recommended to disable all codecs except H265 and H264
  5. In the network section, select SIP INFO and disable IPV6. If you are in a local network, disable ICE/STUN option too.
  6. Open your jitsi conference in the browser (for example : https://jitsi.domain.com/myConference)
  7. Launch a new SIP call from the call menu in Linphone
  8. sip:myConference@KAMAILIO_IP_ADDRESS for direct access to the conference myConference
  9. sip:0@KAMAILIO_IP_ADDRESS for DTMF access (ConfMapper should be configured)