I just install Kamailio 4.4.5 + RTPEngine on Ubuntu Server 16.04
all config copy from https://github.com/whisk/WEBRTC-to-SIP
And using SIP.js latest version to make call between 2 chrome browser.
SIP user register successfully and can using text chat
But when User Invite another user any his Accept call, get this error
Failed to set remote answer sdp: Called with SDP without DTLS fingerprint.
How to solve this issue???
Use sdp_with_transport_like(...) instead of sdp_with_transport(...), because the transport string is UDP/TLS/RTP/SAVPF and the second function is doing an exact match, but the parameter in referred example is only RTP/SAVPF.
Related
I am trying to achieve the multicloud architecture. My network has 2 peers, 1 orderer and a webclient. This network is in Azure. I am trying to add a peer from Google Cloud Platform to the channel of Azure. For this, I created a crypto-config for 3rd peer from Azure webclient. But in the crypto-config, I made the changes like peers in Azure have their own certificates while for the 3rd peer, I placed the newly created certificates. Now I can install, instantiate, invoke and do queries in the peers(1 and 2). And I can install the chaincodes in 3rd peer. But I am unable to instantiate the chaincodes.
Getting the following error: Error: could not assemble transaction, err proposal response was not successful, error code 500, msg error starting container: error starting container: Post http://unix.sock/containers/create?name=dev-(CORE_PEER_ID)-documentCC-1: dial unix /var/run/docker.sock: connect: permission denied
Can anyone guide me on this.
Note: All the peers, orderer, webclient are running in different vm(s)
#soundarya
It doesn’t matter how many places your solution is deployed
The problem is you are running docker by using sudo command try to add docker to sudo group
Below block will help you out
https://www.digitalocean.com/community/questions/how-to-fix-docker-got-permission-denied-while-trying-to-connect-to-the-docker-daemon-socket
To learn more concept about docker.sock
You can refer to my answer in another Can anyone explain docker.sock
I'm trying to implement a basic audio/video-chat functionality through WebRTC by means of ejabberd in JavaScript. To do it, I'm using the Stanza library which implements the following protocols:
XEP-0166: Jingle
XEP-0176: Jingle ICE-UDP Transport Method
And from the docs I know that Ejabberd supports Jingle ICE (XEP-0176). https://docs.ejabberd.im/admin/configuration/#stun-and-turn
When I try sending a Jingle "session-initiate" stanza, as described in XEP-0166, it throws the following error:
<error code='503' type='cancel'>
<service-unavailable xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'/>
<text xml:lang='en' xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'>No module is handling this query</text>
</error>
What version of ejabberd are you using?
18.1
What operating system (version) are you using?
Centos 7
How did you install ejabberd (source, package, distribution)?
Source
If needed, I can post here my installation steps and a config.
Please, advise on how to proceed further with my implementation?
are you trying to send the session-initate to a bare jid (user#host) instead of a full jid (user#host/resource)? The former will be handled by the server and it is quite unlikely that it supports accepting calls that way...
I have implemented FCM for web using fcm documentation.
Everything'll be fine if I set url like : 'http://xxx' I have no error.
But when I set url : 'https://xxx..', I get error:
"Failed to register a ServiceWorker: An SSL certificate error occurred when fetching the script."
code: "messaging/failed-serviceworker-registration"
"Messaging: We are unable to register the default service worker. Failed to register a ServiceWorker: An SSL certificate error occurred when fetching the script. (messaging/failed-serviceworker-registration)."
Can anyone show me how to fix this error?
This is a general problem when wanting to test service workers in a local development environment without proper SSL certificates. It is not specific to Firebase Messaging but pertains to Service Workers in general.
Here is the solution I found when using Google Chrome: Testing Service workers locally with self-signed certificates
Unfortunately, I don't know yet how to circument the issue with other browsers, but probably there must be similar ways.
For Chrome, you need to start a new instance of Chrome, with some flags telling it to ignore SSL certificate errors for your local origin:
In Linux (and maybe Mac):
google-chrome --ignore-certificate-errors --unsafely-treat-insecure-origin-as-secure=https://127.0.0.1 --user-data-dir=/tmp/foo
The https://127.0.0.1 here is the location where your app (and service worker) is hosted locally. You might need to adjust this to use the appropriate port, if serving on a different port than the standard HTTPS port 443, e.g. https://127.0.0.1:3000, when serving your app over HTTPS on port 3000.
The --user-data-dir=/tmp/foo is necessary to start a new instance, with a new user profile, if another instance of Chrome is already running.
In Windows (might vary, depending on where your chrome.exe is):
C:\Program Files (x86)\Google\Chrome\Application\chrome.exe --ignore-certificate-errors --unsafely-treat-insecure-origin-as-secure=https://localhost:1123
Again, you might have to adjust the port.
Easier method that worked for me:
Just paste chrome://flags/#allow-insecure-localhost in your chrome browser, and Enable the setting that says something like "Allow invalid certificates for resources loaded from localhost."
Recently I've been installing upwork application on my debian system.It has installed fine.But when I try to launch it from cli typing: upwork
a bunch of errors happen.
[1008/213534:ERROR:browser_main_loop.cc(173)] Running without the SUID sandbox! See https://code.google.com/p/chromium/wiki/LinuxSUIDSandboxDevelopment for more information on developing with the sandbox on.
[1008/213535:ERROR:renderer_main.cc(200)] Running without renderer sandbox
[1008/213542:ERROR:renderer_main.cc(200)] Running without renderer sandbox
[1008/213542:WARNING:channel.cc(549)] Failed to send message to ack remove remote endpoint (local ID 1, remote ID 1)
[1008/213542:WARNING:channel.cc(549)] Failed to send message to ack remove remote endpoint (local ID 2147483648, remote ID 2)
[1008/213542:ERROR:channel.cc(300)] RawChannel read error (connection broken)
[1008/213542:ERROR:renderer_main.cc(200)] Running without renderer sandbox
[1008/213543:WARNING:channel.cc(549)] Failed to send message to ack remove remote endpoint (local ID 1, remote ID 1)
[1008/213543:WARNING:nss_ssl_util.cc(370)] Unknown SSL error -12218 (SSL_ERROR_ENCRYPTION_FAILURE) mapped to net::ERR_SSL_PROTOCOL_ERROR
[1008/213600:WARNING:nss_ssl_util.cc(370)] Unknown SSL error -12218 (SSL_ERROR_ENCRYPTION_FAILURE) mapped to net::ERR_SSL_PROTOCOL_ERROR
[1008/213600:WARNING:nss_ssl_util.cc(370)] Unknown SSL error -12218 (SSL_ERROR_ENCRYPTION_FAILURE) mapped to net::ERR_SSL_PROTOCOL_ERROR
[1008/214550:WARNING:channel.cc(549)] Failed to send message to ack remove remote endpoint (local ID 1, remote ID 1)
[
How to overcome this errors?
After online searching the decision was found on upwork site forum
unknown ssl error
The problem was how to use the 2 versions of libnss3 packet simultaneously.Libnss3 goes as a security related packet so instead of downgrading it on the system you can simply download the old version of libnss3 compatible with upwork app and then dynamically add a path to this old version to the linker when you need to use upwork app, while the system will have the newest one.
I'm trying to get asterisk 11.20.0 running with WebRTC (sip.js 0.72 which I believe is a fork of jssip), but I'm seeing the following (and the called party rings, but when the phone is answered the call gets hung up).
This is my setup:
What I see:
In the CLI:
[2015-11-24 01:01:53] NOTICE[43619][C-00000002]: res_rtp_asterisk.c:4441 ast_rtp_read: Unknown RTP codec 95 received from '(null)'
In Firefox:
InvalidSessionDescriptionError: Invalid description, no ice-ufrag attribute
Attachments:
SIP Dialogue (Asterisk CLI)
Webphone Log
Config Files (httpd.conf, sip.conf, rtp.conf)
Asterisk Compiled with Libuuid & Friends
What I've tried so far:
Changed webRTC implementations (tried chrome and firefox both with SIPML and SIP.JS)
Set the STUN server to null on the client side (stunServers: ['stun:null'])
Configured properly (I hope) my sip.conf and rtp.conf and httpd.conf
Made sure I have libuuid, uuid and their -devel companions and after i've recompiled asterisk.
What I've read:
http://forums.asterisk.org/viewtopic.php?p=201702
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
http://jssip.net/documentation/misc/interoperability/asterisk/
http://sipjs.com/guides/server-configuration/asterisk/
https://kunjans.wordpress.com/2015/01/09/web-sip-client-sipml5-with-asterisk-13-on-centos-6-6/
http://forums.digium.com/viewtopic.php?f=1&t=89798
Please, if you can, give me a hand. I'm about to smash my box with a sledge hammer.
Faced same issue and followed instruction in http://forums.digium.com/viewtopic.php?f=1&t=90167 realise that:
This issue is caused because you asterisk don't have ICE support, you can solve that by installing the uuid/libuuid and uuid-devel/libuuid-devel packages on your system. Then recompile asterisk(be sure to rerun the configure script before the make command).
I did recompile my Astersik 11.16.0 with patch for ECDH support and fallback to prime256v1 https://issues.asterisk.org/jira/browse/ASTERISK-25265 and looks like lost uuid support at that time. Reverting back to non-patched version (with uuid support, use to be compiled before) resolved my issue with "no ice-ufrag attribute" error in Firefox console and calls are going well now from WebRTC client SIPML5 based to asterisk, but not in opposit direction