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Fullstack VOIP , WebRTC and media Streams Engineer . I solve real world problems . CV : https://altanai.github.io/ Have build secure , fast , enterprise ...
Missing: q= 3A% 2Ftelecom. 2Ftag%
Session Initiation Protocol (SIP) Service Creation Environment (SCE ) for SIP Applications Hosted IP-PBX and its SBC SIP Servlets JAINSLEE What is JAIN SLEE ...
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Specialized in CPaaS, carrier-grade WebRTC-SIP telecom platforms for Unified communication-collaboration, signaling gateways, SBC, soft turrets, ...
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Specialized in CPaaS, carrier-grade WebRTC-SIP telecom platforms for Unified communication-collaboration, signaling gateways, SBC, soft turrets, ...
Missing: 3A% 2Ftelecom. 2Ftag%
VoIP manages Call setup and teardown using IP protocol. The APIs can be used to provide public or internal endpoinst to create mnage calls , conference addon ...
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Me , I'm a simple countryside coder ! :atom: Here are few instances of my work I'm most proud of. Webrtcdevelopment SDK WebRTC development accelerator, MIT.
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It is widley used as aperformnace and load testing tool since it can test SIP equipements like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, and SIP ...
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Code snippet for adding constraints to output media via pipeline and forcing choice of codecs( H264 for video and PCMU for audio ). 1. 2. 3. 4. 5. 6. 7. 8.
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