SIP / RTCP
Download this capture file: ” SIP – RTCP control through NAT Device ”
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SIP is defined by RFC 2543 and is used for multimedia call session setup and control over IP networks. Read more…
This PDF describes the deployment strategy in detail, including planning, deployment and lessons learned. A very useful document if you are planning a large VoIP upgrade or installation. VIEW PDF
This configuration enables a trunk between an Asterisk Open Source PBX and Cisco Unified Call Manager Express: Read more…
Another Aide-Memoir: SIP User Agents in different domains – session establishment (also uses location server): Read more…
Fun numbers are free (no cost) numbers you can call to either test your connection or get several services. The connections to these numbers are usually free of charge, as the “lines” are using the internet to transfer your voice between you and the service. Read more…
Planning for Voice over IP requires an understanding of the various headers added when transporting packetised voice, espcially over an IPSec VPN: Read more…
