VoIP Calls using Nokia Asha SDK 1.1 (27Nov 4pmUTC)

On this webinar is discussed theVoice over IP (VOIP) use within the new Nokia Asha SDK 1.1. As is already knwo the VOIP capatilbity uses the Internet Protocol (IP) to send and receive the voice converted into packets over an IP network.  In order to implement such a functionality the Nokia Asha Platform uses the Session Initiation Protocol (SIP) protocol.

In the Nokia Asha Platform the VOIP platform details are the following:
• Protocols: SIP, RTP/RTCP, STUN
• Supports WiFi internet access
• Call features: Mobile Originating and Mobile Terminating calls
• Hold/Retrieve
• Multi call support (one active, one held, one waiting)
• Call forwarding
• DTMF support

As in almost everything there is a set of capabilities and limitations, the supported capabilities are:
• One media stream (audio) only is supported
• Speech codecs: AMR-WB/NB, G.711 u-law/a-law, G.729, G.726, GSM-EFR/FR.
• Echo cancelation & Dynamic jitter buffer
• Secure call support (SIP TLS + SRTP)
• Network Aaddress Translation /Firewall travelsal: STUN support, NAT keep alive, symmetric signaling/media
• Quality of Service: Media/Signalling QoS marking with Differentiated Service Fields, Bandwidth negotiation, RTP CP Extended Report VoIP metrics
• IPv4 and IPv6 in signaling.
• E.164 numbers


However, the platform has also some restrictions or limitations that are described here:
• Multicast streams are not supported.
• Changing the port number/media type/transport during a session is not supported in the
• Session establishment through two proxies.
• Successful session with proxy failure
• Session through a SIP ALG.
• Offer & Answer: A port number of zero is used to reject offered media for MT sessions.

There are two ways to use the VOIP within Nokia Asha. One is natively and other is through Java as integrated within an application.

Reference:
Webinar presentation