Please download our latest update of WebRTC SIP SDK.
Change log:
- outgoing call connection timeout reduced
- custom headers for invite messages made possible
- customizable user-agent SIP field added
- DTMF implemented
- possibility to perform multiple simultaneous calls added
- video/audio recording implemented
- application interface changes (done more consistent, obsolete methods removed, parameters changed, renamings)
- possibility to select camera and microphone added
- tag added to TO SIP header field
- customizable register expiration timeout
- some of the deprecation warnings resolved
- hold redone to be more flexible with different servers
- various fixes
- new developer manual written