New features of the VoIP SIP Client SDK:
. g729 and g723 CodecÃÂ´s support. Multiple and single Codec selection support. Failure codes support (get SIP Message Response Code, SIP Message Response Text). RTP/RTCP Port setting (for inbound RTP traffic). Reduce audio latency and audio latency settings (properties: MinPrefetchCount, MaxPrefetchCount, MaxRTPPackets). Media status (Events: OnLocalMediaStarted, OnLocalMediaStoped, OnRemoteMediaStarted, OnRemoteMediaStoped). Get used codec per line. Custom Ringtone (play wav) support (property: RingtoneFile). Play wav to a selected phone line (methods: StartPlayingAtLine, StopPlayingAtLine). Redirect Call to other phone line. Load and Save Configurations (methods: LoadConfiguration, StoreConfiguration). Complete new, re-written and updated samples with source code. and much more!
Here is a list of the main features of the VoIP SIP Client SDK:
. Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider. VoIP conferencing with crystal clear sound even for both low and high-bandwidth users G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10, iLBC, L16 and g729 & g723 Codec. Open standards-based and interoperable with all o
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