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…
Another aide-memoir, these are the setting required to enable a sipgate account (and PSTN number) to connect to a phone registered to your Elastix / Asterisk PBX:
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Attacks on Asterisk-based telephony systems are not uncommon. This video explains how to mitigate some attack vectors:
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Another aide-memoir reproduced from voip-info.org: Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny. SEE ALSO: The Asterisk Book
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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
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…