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When developing SIP software or deploying a new SIP infrastructure, it is important to test the capability of servers and IP networks to handle certain call load: number of concurrent calls and number of calls per second.
The introduction of Session Initiation Protocol (SIP) means that point-to-point communications are no longer restricted to voice calls but can now be extended to multimedia technologies such as video. IVR manufacturers have extended their systems into IVVR (interactive voice and video response), especially for the mobile phone networks.
SIP requests and responses may be generated by any SIP user agent; user agents are divided into clients (UACs), which initiate requests, and servers (UASes), which respond to them. [ 1 ] : §8 A single user agent may act as both UAC and UAS for different transactions: [ 1 ] : p26 for example, a SIP phone is a user agent that will be a UAC when ...
Codenomicon Defensics: commercial test automation framework; Ixia (company) commercial SIP-VoIP and Video test and emulation and load test platform; Mu Dynamics: commercial SIP-VoIP, RTSP-IPTV Triple Play service assurance platform
Jami (based on DHT and SIP) Savoir-faire Linux Inc. 2002 August Open Standard: 40-digit address Yes Yes Yes Yes No Yes Medium Yes Yes Yes Yes No Yes ? Jami (based on DHT and SIP) Matrix: Matrix.org 2014 Sep [11] [failed verification] Open standard @Username:Hostname (MXID) Yes Yes, mandatory Yes, default for private conversations [12] Yes Yes Yes
Usually, ITSPs negotiate agreements for route termination to various parts of the world from multiple VoIP providers. ITSP customers may be able to choose a VoIP provider for their VoIP calls. Customers may be able to do this by specifying the maximum price they are willing to pay per minute for a call and the lowest quality they are willing to ...
SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities. [1]
oSIP is a free software library for VoIP applications implementing lower layers of Session Initiation Protocol (SIP). The library includes the minimal codebase required by any SIP application and offers enough flexibility to implement any SIP extension or behavior.
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