FIx missing ACK issue with webrtc gateway

dOpensource / dsiprouter

UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services

https://dsiprouter.org

Apache License 2.0

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Open devopsec opened 8 months ago

devopsec commented 8 months ago

webrtc to SIP, and vice versa translation

devopsec commented 8 months ago

currently the hurdles to tackle are:

asterisk SIP phone to external webrtc client the ACK is not forwarded to the SIP phone