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Hi, I'm using SIP as RTMP, but would like to be able to "connect to the conversation" and input audio/speak. For example; is there a way to connect to a SIP as RTMP call using an webrtc socket? Any tips are welcome. Thanks in advance.

Hi, When try to call a number from a conference call provider, it is stuck on TRYING. The call is never established. In the logs I see this: <-------------------- SIP/2.0 100 Trying from: xxxxxxx to: xxxxxxx time: 1501228599418 timeStamp: isSender: false transactionId...