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Hello Everyone.
I am looking for a library (library for Java or C # or Vb.net) that will let me call a number through a VoIP service

I am interested in a really small library because I just need to call and hear the voice of the receiver using this data for call setup
SIP server (or proxy, or domain)	sip.sipserver.com
SIP proxy (or "Outbound Proxy")	        leave blank
STUN server	                        stun.sipserver.com
Username (or User ID)	                Myusername
Password	                        Mypassword
Auth name (or Auth ID)	                Myusername
Display Name	                        My name
Register (or Send registration request)	Yes
G729a Codec Name (for buggy Linksys/Sipura/Cisco ATAs)	G729.
The default codec name in those adapters is set to non RFC compliant "G729a" and might not work with our service, go to Admin/Advanced/SIP menu in ATA settings to change the codec name.
Registry Expiry (or Registration interval)   120 sec (2 minutes) if your SIP client is behind NAT router.


If there isn't a library like that, or are too complex for my goal, maybe Java or C # have built something that allows me to hear the voice of the person I'm calling?
Posted 24-Mar-12 1:59am
ludo237311
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Solution 1

SIP is still an evolving standard. There is an active SIPForum (www.sipforum.org) group that is currently developing SIPConnect 1.1 to standardize on an interface between a SIP Service provider (e.g. Vonage) and a SIP-PBX.

There is an activity called BLISS for defining the "best practices" to implement SIP features between User Agents. BLISS reference: http://www.bliss-ietf.org/charter.html[^]
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ludo237 24-Mar-12 8:52am
   
Thank you i will try it this afternoon :)
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Solution 2

I have been working with the Ozeki VOIP SIP SDK. http://www.voip-sip-sdk.com/[^] and the SDK from http://www.portsip.com/[^]

Thus far, using the Ozeki product, I have created a SoftPhone application. I have, also, created a Dialer that distinguishes between a Human answer vs an Answering machine. When a Human answers, the dialer does a Blind Transfer to a CSR extension.

I prefer the PortSip IVR SDK to create a dynamic IVR.

I am utilizing this with an Asterisk PBX running on Linux.
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