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2. Comparison of VoIP software - Wikipedia

   en.wikipedia.org/wiki/Comparison_of_VoIP_software

   This is a comparison of voice over IP (VoIP) software that examines applications and systems used for conducting voice and multimedia communications across Internet Protocol (IP) networks. VoIP technology has transformed telecommunications by offering alternatives to traditional telephony systems while providing enhanced features and cost savings.

3. SIMPLE (instant messaging protocol) - Wikipedia

   en.wikipedia.org/wiki/SIMPLE_(instant_messaging...

   SIMPLE, the Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions, is an instant messaging (IM) and presence protocol suite based on Session Initiation Protocol (SIP) managed by the Internet Engineering Task Force.

4. Session Initiation Protocol - Wikipedia

   en.wikipedia.org/wiki/Session_Initiation_Protocol

   SIP performance tester software is used to simulate SIP and RTP traffic to see if the server and IP network are stable under the call load. [24] The software measures performance indicators like answer delay, answer/seizure ratio , RTP jitter and packet loss , round-trip delay time .

5. List of SIP software - Wikipedia

   en.wikipedia.org/wiki/List_of_SIP_software

   Intertex SIP transparent routers, firewalls and ADSL modems, for broadband deployments and SOHO market; Juniper Networks Netscreen and SRX firewalls include complete SIP Application Layer Gateway support; Linux Netfilter's SIP conntrack helper fully understands SIP and can classify (for QOS) and NAT all related traffic; Netopia Netopia supports ALG

6. SIP extensions for the IP Multimedia Subsystem - Wikipedia

   en.wikipedia.org/wiki/SIP_extensions_for_the_IP...

   The Session Initiation Protocol (SIP) is the signaling protocol selected by the 3rd Generation Partnership Project (3GPP) [1] [2] to create and control multimedia sessions with multiple participants in the IP Multimedia Subsystem (IMS). It is therefore a key element in the IMS framework.

7. VoIP phone - Wikipedia

   en.wikipedia.org/wiki/VoIP_phone

   STUN is used in some of the sip phones to enable the SIP/RTP packets to cross boundaries of two different IP networks. A packet becomes unroutable between two sip elements if one of the networks uses private IP address range and other is in public IP address range. Stun is a mechanism to enable this border traversal.