Call functions like mute, conference, hold, transfer, and call recording should be available. These options should be visible during a call. When you transfer a call, it will ask for your permission to move the call to another person first.
Custom ringtones
Echo cancellation and noise suppression
Pop-up notifications
Presence/online status
Call history
Voicemail messages and missed calls notifications
Contact integration from LDAP, Outlook, or CSV
Low resource consumption
Supports SIP, XMPP, and IAX accounts
TLS, SRTP, and ZRTP encryption
Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow
Voice Codecs: H264 (paid), VP8
RAW SIP technology, not WebRTC.
Similar Projects: Zoiper SIP Webphone
Hi there,
I've read your project description and I am confident enough that I can handle this project according to your expectations. I have done similar projects before and I want to take over this project as well. If you're interested then please contact me to see my portfolio :) I'll be waiting for your response.
Regards
We are an exceptionally competent team with an average experience of 10+ years in various fields of Information Technology. We had an opportunity to work on some of the rare and highly skilled areas of telecom protocols like SS7,GSM-MAP,Megaco, Asterisk, PBX, RADIUS etc. We have clients across globe primarily from US and Spain.
Price quoted is ballpark. Please share your availability to discuss about your project.
Greetings, I can build your application as you mentioned in project description. I am an independent and professional developer holding 100% completion rate. I am highly interested about your project, Please knock me for a quick discuss