srtp | A Go implementation of SRTP | Encryption library

I'm trying to launch this on AWS Ubuntu.

It works fine under Chrome on localhost. (There was an issue with Firefox, hopefully running remotely with HTTPS will make the problem disappear. But that's unrelated to this question.)

I opened the ports that are specified on readme.MD using the AWS console (inbound TCP to port 3000, inbound UDP to ports 40000-49999, all outgoing traffic is allowed.)

Then adapted config.json to:

I have a swift project that uses the GoogleWebRTC pod.

When trying to negotiate the OPUS codec for audio calls i find that the peer connection is successfully setup, however i am experiencing one way audio. SRTP is being sent from my iPhone to the other party successfully, and SRTP is being sent from the other party to my iPhone, however my phone/app is not playing the incoming SRTP to the user. If i negotiate any other codec (G722 for example) then i get 2 way audio, it's just when i try to negotiate OPUS that i don't hear any incoming audio on my iPhone.

Couldn't see anything relevant in the logs, but looking for some pointers on how to troubleshoot this or what could potentially be the cause of this issue.

I'm using the google WebRTC iOS SDK.

Here is the code in my webrtc class where i initialize the audio session if that helps.

What type of data protocol is inside ChannelData when webrtc completely uses turn relay? Does not look like SRTP;

I'm trying to cross compile aws webrtc library for arm but I encountered an error while linking C excutable test_srtp. I mixed the instruction of Readme.md and some stackoverflow post for the command. But I am still stuck here for 1 week and I can not find the case on google. How can I fix it?

What command I input

Apologies as the question might be a bit vague. Trying to establish webrtc connection to a webrtc-gateway. While performing the dtls handshaking with accept or connect function, it is throwing SocketException.

Here is the error:

I am trying to learn more about inline and SRTP. Unfortunately I don't have the understanding to make this sample more minimal, but it's not too big:

I am developing a multi-account SIP softphone with custom media transport. I need to manage multiple calls simultaneously. I have read in "PJSUA2 Book" (3 PJSUA2-High Level API / General concepts / Asynchronous Operations):

(...) all operations that involve sending and receiving SIP messages are asynchronous, meaning that the function that invokes the operation will complete immediately (...) When this function [pj::Call::makeCall] returns successfully, it does not mean that the call has been established, but rather it means that the call has been initiated successfully. You will be given the report of the call progress and/or completion in the onCallState() callback method of Call class.

So I thought that I could call pj::Call::makeCall whenever/wherever I needed to initiate a call, the function would return nearly immediately and any progress would be reported in the callback methods. But today in my test environment I noticed a few seconds delay between the moment I called the function and it returned. Below there are detailed PJSIP logs that appear between makeCall is called and it returned. One can see almost 4s gap before "RTP socket reachable":

env: node: No such file or directory

I checked if my directory for node wasn't wrong and it's fine.

I tried these following answers already: 1. https://github.com/nodejs/node-v0.x-archive/issues/3911 2. https://github.com/creationix/nvm/issues/1702 3. browserify error /usr/bin/env: node: No such file or directory

I loved the idea used in the accepted answer here. My understanding of it is that it is an elegant way of using an interface as a method parameter without having to implement the interface. SRTP is used for duck typing to adopt a type (excuse the OO term, happy to learn the FP one) to the interface: