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< h1 id = "overview" > < a class = "header" href = "#overview" > Overview< / a > < / h1 >
< p > This document explains how to enable VoIP relaying on your homeserver with
TURN.< / p >
< p > The synapse Matrix homeserver supports integration with TURN server via the
< a href = "https://tools.ietf.org/html/draft-uberti-behave-turn-rest-00" > TURN server REST API< / a > . This
allows the homeserver to generate credentials that are valid for use on the
TURN server through the use of a secret shared between the homeserver and the
TURN server.< / p >
< p > The following sections describe how to install < a href = "https://github.com/coturn/coturn" > coturn< / a > (which implements the TURN REST API) and integrate it with synapse.< / p >
< h2 id = "requirements" > < a class = "header" href = "#requirements" > Requirements< / a > < / h2 >
< p > For TURN relaying with < code > coturn< / code > to work, it must be hosted on a server/endpoint with a public IP.< / p >
< p > Hosting TURN behind NAT requires port forwaring and for the NAT gateway to have a public IP.
However, even with appropriate configuration, NAT is known to cause issues and to often not work.< / p >
< h2 id = "coturn-setup" > < a class = "header" href = "#coturn-setup" > < code > coturn< / code > setup< / a > < / h2 >
< h3 id = "initial-installation" > < a class = "header" href = "#initial-installation" > Initial installation< / a > < / h3 >
< p > The TURN daemon < code > coturn< / code > is available from a variety of sources such as native package managers, or installation from source.< / p >
< h4 id = "debian-installation" > < a class = "header" href = "#debian-installation" > Debian installation< / a > < / h4 >
< p > Just install the debian package:< / p >
< pre > < code class = "language-sh" > apt install coturn
< / code > < / pre >
< p > This will install and start a systemd service called < code > coturn< / code > .< / p >
< h4 id = "source-installation" > < a class = "header" href = "#source-installation" > Source installation< / a > < / h4 >
< ol >
< li >
< p > Download the < a href = "https://github.com/coturn/coturn/releases/latest" > latest release< / a > from github. Unpack it and < code > cd< / code > into the directory.< / p >
< / li >
< li >
< p > Configure it:< / p >
< pre > < code class = "language-sh" > ./configure
< / code > < / pre >
< p > You may need to install < code > libevent2< / code > : if so, you should do so in
the way recommended by your operating system. You can ignore
warnings about lack of database support: a database is unnecessary
for this purpose.< / p >
< / li >
< li >
< p > Build and install it:< / p >
< pre > < code class = "language-sh" > make
make install
< / code > < / pre >
< / li >
< / ol >
< h3 id = "configuration" > < a class = "header" href = "#configuration" > Configuration< / a > < / h3 >
< ol >
< li >
< p > Create or edit the config file in < code > /etc/turnserver.conf< / code > . The relevant
lines, with example values, are:< / p >
< pre > < code > use-auth-secret
static-auth-secret=[your secret key here]
realm=turn.myserver.org
< / code > < / pre >
< p > See < code > turnserver.conf< / code > for explanations of the options. One way to generate
the < code > static-auth-secret< / code > is with < code > pwgen< / code > :< / p >
< pre > < code class = "language-sh" > pwgen -s 64 1
< / code > < / pre >
< p > A < code > realm< / code > must be specified, but its value is somewhat arbitrary. (It is
sent to clients as part of the authentication flow.) It is conventional to
set it to be your server name.< / p >
< / li >
< li >
< p > You will most likely want to configure coturn to write logs somewhere. The
easiest way is normally to send them to the syslog:< / p >
< pre > < code class = "language-sh" > syslog
< / code > < / pre >
< p > (in which case, the logs will be available via < code > journalctl -u coturn< / code > on a
systemd system). Alternatively, coturn can be configured to write to a
logfile - check the example config file supplied with coturn.< / p >
< / li >
< li >
< p > Consider your security settings. TURN lets users request a relay which will
connect to arbitrary IP addresses and ports. The following configuration is
suggested as a minimum starting point:< / p >
< pre > < code > # VoIP traffic is all UDP. There is no reason to let users connect to arbitrary TCP endpoints via the relay.
no-tcp-relay
# don't let the relay ever try to connect to private IP address ranges within your network (if any)
# given the turn server is likely behind your firewall, remember to include any privileged public IPs too.
denied-peer-ip=10.0.0.0-10.255.255.255
denied-peer-ip=192.168.0.0-192.168.255.255
denied-peer-ip=172.16.0.0-172.31.255.255
# recommended additional local peers to block, to mitigate external access to internal services.
# https://www.rtcsec.com/article/slack-webrtc-turn-compromise-and-bug-bounty/#how-to-fix-an-open-turn-relay-to-address-this-vulnerability
no-multicast-peers
denied-peer-ip=0.0.0.0-0.255.255.255
denied-peer-ip=100.64.0.0-100.127.255.255
denied-peer-ip=127.0.0.0-127.255.255.255
denied-peer-ip=169.254.0.0-169.254.255.255
denied-peer-ip=192.0.0.0-192.0.0.255
denied-peer-ip=192.0.2.0-192.0.2.255
denied-peer-ip=192.88.99.0-192.88.99.255
denied-peer-ip=198.18.0.0-198.19.255.255
denied-peer-ip=198.51.100.0-198.51.100.255
denied-peer-ip=203.0.113.0-203.0.113.255
denied-peer-ip=240.0.0.0-255.255.255.255
# special case the turn server itself so that client-> TURN-> TURN-> client flows work
# this should be one of the turn server's listening IPs
allowed-peer-ip=10.0.0.1
# consider whether you want to limit the quota of relayed streams per user (or total) to avoid risk of DoS.
user-quota=12 # 4 streams per video call, so 12 streams = 3 simultaneous relayed calls per user.
total-quota=1200
< / code > < / pre >
< / li >
< li >
< p > Also consider supporting TLS/DTLS. To do this, add the following settings
to < code > turnserver.conf< / code > :< / p >
< pre > < code > # TLS certificates, including intermediate certs.
# For Let's Encrypt certificates, use `fullchain.pem` here.
cert=/path/to/fullchain.pem
# TLS private key file
pkey=/path/to/privkey.pem
# Ensure the configuration lines that disable TLS/DTLS are commented-out or removed
#no-tls
#no-dtls
< / code > < / pre >
< p > In this case, replace the < code > turn:< / code > schemes in the < code > turn_uris< / code > settings below
with < code > turns:< / code > .< / p >
< p > We recommend that you only try to set up TLS/DTLS once you have set up a
basic installation and got it working.< / p >
< p > NB: If your TLS certificate was provided by Let's Encrypt, TLS/DTLS will
not work with any Matrix client that uses Chromium's WebRTC library. This
currently includes Element Android & iOS; for more details, see their
< a href = "https://github.com/vector-im/element-android/issues/1533" > respective< / a >
< a href = "https://github.com/vector-im/element-ios/issues/2712" > issues< / a > as well as the underlying
< a href = "https://bugs.chromium.org/p/webrtc/issues/detail?id=11710" > WebRTC issue< / a > .
Consider using a ZeroSSL certificate for your TURN server as a working alternative.< / p >
< / li >
< li >
< p > Ensure your firewall allows traffic into the TURN server on the ports
you've configured it to listen on (By default: 3478 and 5349 for TURN
traffic (remember to allow both TCP and UDP traffic), and ports 49152-65535
for the UDP relay.)< / p >
< / li >
< li >
< p > If your TURN server is behind NAT, the NAT gateway must have an external,
publicly-reachable IP address. You must configure coturn to advertise that
address to connecting clients:< / p >
< pre > < code > external-ip=EXTERNAL_NAT_IPv4_ADDRESS
< / code > < / pre >
< p > You may optionally limit the TURN server to listen only on the local
address that is mapped by NAT to the external address:< / p >
< pre > < code > listening-ip=INTERNAL_TURNSERVER_IPv4_ADDRESS
< / code > < / pre >
< p > If your NAT gateway is reachable over both IPv4 and IPv6, you may
configure coturn to advertise each available address:< / p >
< pre > < code > external-ip=EXTERNAL_NAT_IPv4_ADDRESS
external-ip=EXTERNAL_NAT_IPv6_ADDRESS
< / code > < / pre >
< p > When advertising an external IPv6 address, ensure that the firewall and
network settings of the system running your TURN server are configured to
accept IPv6 traffic, and that the TURN server is listening on the local
IPv6 address that is mapped by NAT to the external IPv6 address.< / p >
< / li >
< li >
< p > (Re)start the turn server:< / p >
< ul >
< li >
< p > If you used the Debian package (or have set up a systemd unit yourself):< / p >
< pre > < code class = "language-sh" > systemctl restart coturn
< / code > < / pre >
< / li >
< li >
< p > If you installed from source:< / p >
< pre > < code class = "language-sh" > bin/turnserver -o
< / code > < / pre >
< / li >
< / ul >
< / li >
< / ol >
< h2 id = "synapse-setup" > < a class = "header" href = "#synapse-setup" > Synapse setup< / a > < / h2 >
< p > Your homeserver configuration file needs the following extra keys:< / p >
< ol >
< li > " < code > turn_uris< / code > " : This needs to be a yaml list of public-facing URIs
for your TURN server to be given out to your clients. Add separate
entries for each transport your TURN server supports.< / li >
< li > " < code > turn_shared_secret< / code > " : This is the secret shared between your
homeserver and your TURN server, so you should set it to the same
string you used in turnserver.conf.< / li >
< li > " < code > turn_user_lifetime< / code > " : This is the amount of time credentials
generated by your homeserver are valid for (in milliseconds).
Shorter times offer less potential for abuse at the expense of
increased traffic between web clients and your homeserver to
refresh credentials. The TURN REST API specification recommends
one day (86400000).< / li >
< li > " < code > turn_allow_guests< / code > " : Whether to allow guest users to use the
TURN server. This is enabled by default, as otherwise VoIP will
not work reliably for guests. However, it does introduce a
security risk as it lets guests connect to arbitrary endpoints
without having gone through a CAPTCHA or similar to register a
real account.< / li >
< / ol >
< p > As an example, here is the relevant section of the config file for < code > matrix.org< / code > . The
< code > turn_uris< / code > are appropriate for TURN servers listening on the default ports, with no TLS.< / p >
< pre > < code > turn_uris: [ " turn:turn.matrix.org?transport=udp" , " turn:turn.matrix.org?transport=tcp" ]
turn_shared_secret: " n0t4ctuAllymatr1Xd0TorgSshar3d5ecret4obvIousreAsons"
turn_user_lifetime: 86400000
turn_allow_guests: True
< / code > < / pre >
< p > After updating the homeserver configuration, you must restart synapse:< / p >
< ul >
< li > If you use synctl:
< pre > < code class = "language-sh" > # Depending on how Synapse is installed, synctl may already be on
# your PATH. If not, you may need to activate a virtual environment.
synctl restart
< / code > < / pre >
< / li >
< li > If you use systemd:
< pre > < code class = "language-sh" > systemctl restart matrix-synapse.service
< / code > < / pre >
< / li >
< / ul >
< p > ... and then reload any clients (or wait an hour for them to refresh their
settings).< / p >
< h2 id = "troubleshooting" > < a class = "header" href = "#troubleshooting" > Troubleshooting< / a > < / h2 >
< p > The normal symptoms of a misconfigured TURN server are that calls between
devices on different networks ring, but get stuck at " call
connecting" . Unfortunately, troubleshooting this can be tricky.< / p >
< p > Here are a few things to try:< / p >
< ul >
< li >
< p > Check that you have opened your firewall to allow TCP and UDP traffic to the
TURN ports (normally 3478 and 5349).< / p >
< / li >
< li >
< p > Check that you have opened your firewall to allow UDP traffic to the UDP
relay ports (49152-65535 by default).< / p >
< / li >
< li >
< p > Try disabling < code > coturn< / code > 's TLS/DTLS listeners and enable only its (unencrypted)
TCP/UDP listeners. (This will only leave signaling traffic unencrypted;
voice & video WebRTC traffic is always encrypted.)< / p >
< / li >
< li >
< p > Some WebRTC implementations (notably, that of Google Chrome) appear to get
confused by TURN servers which are reachable over IPv6 (this appears to be
an unexpected side-effect of its handling of multiple IP addresses as
defined by
< a href = "https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-12" > < code > draft-ietf-rtcweb-ip-handling< / code > < / a > ).< / p >
< p > Try removing any AAAA records for your TURN server, so that it is only
reachable over IPv4.< / p >
< / li >
< li >
< p > If your TURN server is behind NAT:< / p >
< ul >
< li >
< p > double-check that your NAT gateway is correctly forwarding all TURN
ports (normally 3478 & 5349 for TCP & UDP TURN traffic, and 49152-65535 for the UDP
relay) to the NAT-internal address of your TURN server. If advertising
both IPv4 and IPv6 external addresses via the < code > external-ip< / code > option, ensure
that the NAT is forwarding both IPv4 and IPv6 traffic to the IPv4 and IPv6
internal addresses of your TURN server. When in doubt, remove AAAA records
for your TURN server and specify only an IPv4 address as your < code > external-ip< / code > .< / p >
< / li >
< li >
< p > ensure that your TURN server uses the NAT gateway as its default route.< / p >
< / li >
< / ul >
< / li >
< li >
< p > Enable more verbose logging in coturn via the < code > verbose< / code > setting:< / p >
< pre > < code > verbose
< / code > < / pre >
< p > ... and then see if there are any clues in its logs.< / p >
< / li >
< li >
< p > If you are using a browser-based client under Chrome, check
< code > chrome://webrtc-internals/< / code > for insights into the internals of the
negotiation. On Firefox, check the " Connection Log" on < code > about:webrtc< / code > .< / p >
< p > (Understanding the output is beyond the scope of this document!)< / p >
< / li >
< li >
< p > You can test your Matrix homeserver TURN setup with < a href = "https://test.voip.librepush.net/" > https://test.voip.librepush.net/< / a > .
Note that this test is not fully reliable yet, so don't be discouraged if
the test fails.
< a href = "https://github.com/matrix-org/voip-tester" > Here< / a > is the github repo of the
source of the tester, where you can file bug reports.< / p >
< / li >
< li >
< p > There is a WebRTC test tool at
< a href = "https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/" > https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/< / a > . To
use it, you will need a username/password for your TURN server. You can
either:< / p >
< ul >
< li >
< p > look for the < code > GET /_matrix/client/r0/voip/turnServer< / code > request made by a
matrix client to your homeserver in your browser's network inspector. In
the response you should see < code > username< / code > and < code > password< / code > . Or:< / p >
< / li >
< li >
< p > Use the following shell commands:< / p >
< pre > < code class = "language-sh" > secret=staticAuthSecretHere
u=$((`date +%s` + 3600)):test
p=$(echo -n $u | openssl dgst -hmac $secret -sha1 -binary | base64)
echo -e " username: $u\npassword: $p"
< / code > < / pre >
< p > Or:< / p >
< / li >
< li >
< p > Temporarily configure coturn to accept a static username/password. To do
this, comment out < code > use-auth-secret< / code > and < code > static-auth-secret< / code > and add the
following:< / p >
< pre > < code > lt-cred-mech
user=username:password
< / code > < / pre >
< p > < strong > Note< / strong > : these settings will not take effect unless < code > use-auth-secret< / code >
and < code > static-auth-secret< / code > are disabled.< / p >
< p > Restart coturn after changing the configuration file.< / p >
< p > Remember to restore the original settings to go back to testing with
Matrix clients!< / p >
< / li >
< / ul >
< p > If the TURN server is working correctly, you should see at least one < code > relay< / code >
entry in the results.< / p >
< / li >
< / ul >
< / main >
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