I'm trying to connect webrtc with my asterisk server. Using Asterisk 12.8 and simpl5. I hear no sound on both sides. Wireshark shows udp packets coming from asterisk but no packets from browser. What can I do?
Check your firewall and NAT settings of Asterisk. Go through the sip debug logs.
Related
I'm writing my own simple soft phone, but I fail to understand the basics of how SIP works, so I wanted to see on a low level how a server responds to messages like the REGISTER request. I have a simple OpenWRT PBX in my router at home, and a couple of hard and Android softphones are working normally - they seem to register, perform calls both between themselves and outside too, so I'm sure it's functional. It's serving requests over UDP and port 5060.
But when I try to get a response from the server on a low level, how I used to do with tcp and telnet, just to see what's going on when a phone is working, the server doesn't seem to respond anything.
Can you please guide me how I can use netcat or some other telnet-like tool to see how the server responds over udp to any request, be them good or malformed?
Use tcpdump tool over telnet
Tcpdump Commands – A Network Sniffer Tool
check the link below.
https://www.tecmint.com/12-tcpdump-commands-a-network-sniffer-tool/
The good tool is Wireshark very good support for SIP message flow and RTP/RTCP.
You may capture the packets from the OpenWRT PBX and from your development device in order to compare things. Good learning!
I have build a one to one webrtc communication system (peer to peer). I use coturn as a STUN and as TURN.
The connection works in chrome (72) and firefox (65) whithot VPN.
The connection also works in chrome and firefox, when behind a symetric NAT (for example while using a mobile-phone).
The connection works also when using VPN and chrome.
The connection (ICE) failed, when using VPN and firefox!
Testet with a couple of popular VPN's.
Any Ideas, what could be wrong/ how to solve?
My TURN-Server Configuration looks like:
listening-port=3478
tls-listening-port=5349
realm=xyz.com
min-port=49152
max-port=65535
user=uuuuuuuuuuuuu:ppppppppppp
pkey=/usr/local/psa/var/modules/letsencrypt/etc/live/turn.xyz.net/privkey.pem
cert=/usr/local/psa/var/modules/letsencrypt/etc/live/turn.xyz.net/cert.pem
CA-file=/usr/local/psa/var/modules/letsencrypt/etc/live/turn.xyz.net/chain.pem
mysql-userdb="host=localhost dbname=turnxxxxxx user=uuuuuuuuuuuu password=ppppppppppp port=3306"
fingerprint
lt-cred-mech
You should expect any WebRTC-related application to fail in all browsers, when VPN is on. I have exactly opposite situation - when VPN is on, no browser works except FireFox.
And in my app no TURN servers are used at all.
Somehow my VPN misses FireFox.
In general, WebRTC is prohibited by VPNs, because of IP address leak paranoia. Your real IP address can leak through signaling messages and STUN binding response, so once VPNs see STUN binding requests, they just kill it. VPNs cannot kill your signaling messages so your private IP can still leak, but that's a problem only with IPV6, where private and public IPs are the same. But, when it's time for STUN binding request/respose, VPNs kill it. Not sure how your Chrome survives.
I am decrypting network encrypted packets of a test device. I am following the youtube video below, and have included the content of my key file. I am unable to decrypt.
Video: https://www.youtube.com/watch?v=vQtur8fqErI
Keyfile content: https://wiki.wireshark.org/SampleCaptures#SSL_with_decryption_keys
My set up is like this;
I have apple tv connected to a Cisco switch with port mirroring capability. I have my test laptop connected to the same switch with Wireshark installed. I have set up port mirroring in the switch to send network packets of Apple TV to my laptop's port. I start Wireshark to watch all calls. I can see all network calls, but the ones are encrypted not decrypting. I have added the key file as per the video but still not working.
I would really appreciate anyone's help. Thanks all.
I'm using Game Maker: Studio and already got the UDP broadcasting to work, but only for people using the internet in the same router. But, I want to broadcast via Hamachi too, so there's no need to enter the IP manually.
This is the function I can use: http://docs.yoyogames.com/source/dadiospice/002_reference/networking/network_send_udp.html
And this one I use for local broadcast: http://docs.yoyogames.com/source/dadiospice/002_reference/networking/network_send_broadcast.html
Your gamemaker UDP packets are as good as any other. So if you can send ping requests to an IP, you should be able to send it any UDP packet you want.
When you ping an address through hamachi, you are actually sending UDP packets to it. So if you want to know if UDP communication will work through hamachi, try to ping someone with the console (type "ping [his IP]"). If it works, then any UDP packet with he correct address should travel correctly, including the ones emitted by your gamemaker app.
Be aware, though, that you may have to cope with a greater packet loss ratio than on your local network. Contrarily to TCP, UDP packets may sometimes not reach their destination.
I have developed a p2p video chat using Webrtc. I am aware that STUN or TURN server is required to identify the public IP behind NAT. Currently am using Google's STUN server.
I have the application installed in the server connected to the LAN which will not have internet access, do I need to install the STUN server in my server to make Webrtc video chat work within the LAN?
Peers should be able to connect within a LAN (on the same side of a NAT) without STUN, i.e. using the host candidates. Try it out!
In general, you won't need a STUN server. However, depending on the firewall configuration, you may actually need STUN (and even TURN). For example, at the SFHTML5 WebRTC Hackathon, we were on a corporate guest WiFi network that blocked local UDP and TCP traffic. Making successful calls required a TURN server in this case.