SIP Client
From Yate Documentation
(Difference between revisions)
(Created page with "SIP client requirements: - [http://yateclient.yate.ro YateClient] * Voice - implemented * Video - soon to be implemented * All free codecs - gsm, speex, alaw, mulaw * Regist...") |
Revision as of 16:45, 24 October 2012
SIP client requirements: - YateClient
- Voice - implemented
- Video - soon to be implemented
- All free codecs - gsm, speex, alaw, mulaw
- Registration - implemented
- Autentification to multiple endpoints - implemented
- Multiple lines (channels) - implemented
- Voicemail - depending on the service
- Hold - implemented
- Transfer - implemented
- Conference - implemented
- ENUM - soon to be implemented
- Dialing using URI - implemented
- Smart routing - low cost routing via multiple providers - implemented
- TFTP provisioning - soon to be implemented
- Local address book
