Last updated: 2026-06-25

Asterisk TLS/SSL Configuration Guide

This guide provides recommended TLS/SSL settings for Asterisk PBX to encrypt SIP signaling traffic. Securing VoIP communications with TLS prevents eavesdropping on call setup and authentication credentials.

Prerequisites#

• Asterisk 20 or later (with PJSIP, which is the recommended SIP channel driver); Asterisk 22 is the current LTS and Asterisk 20 is nearing the end of its full support window
• OpenSSL 1.1.1 or later
• SSL certificates (server certificate, private key, and CA certificate)

Certificate Setup#

Asterisk needs a certificate and private key for TLS. You can use certificates from a CA like Let's Encrypt, or generate self-signed certificates for internal use.

Combine your certificate and private key if needed (Asterisk often expects a combined PEM file):

cat /etc/asterisk/keys/cert.pem /etc/asterisk/keys/privkey.pem > /etc/asterisk/keys/asterisk.pem
chmod 640 /etc/asterisk/keys/asterisk.pem
chown root:asterisk /etc/asterisk/keys/asterisk.pem

PJSIP TLS Transport Configuration#

PJSIP is the modern SIP channel driver in Asterisk and is recommended over the legacy chan_sip. Configure a TLS transport in pjsip.conf.

Transport Settings#

[transport-tls]
type = transport
protocol = tls
bind = 0.0.0.0:5061

; Certificate files
cert_file = /etc/asterisk/keys/asterisk.pem
priv_key_file = /etc/asterisk/keys/privkey.pem
ca_list_file = /etc/asterisk/keys/ca.crt

; TLS method - tlsv1_2 sets the minimum version
; When using OpenSSL 1.1.1+, TLS 1.3 is also available
method = tlsv1_2

; Cipher suites (OpenSSL format)
cipher = ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305

; Verify remote certificates
verify_server = yes
verify_client = no

A PJSIP transport binds to a single address family, so a transport bound to 0.0.0.0 listens on IPv4 only. For dual-stack, define a second TLS transport bound to [::] for IPv6:

[transport-tls-v6]
type = transport
protocol = tls
bind = [::]:5061

; Certificate files
cert_file = /etc/asterisk/keys/asterisk.pem
priv_key_file = /etc/asterisk/keys/privkey.pem
ca_list_file = /etc/asterisk/keys/ca.crt

method = tlsv1_2
cipher = ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305
verify_server = yes
verify_client = no

Configuration Explained#

• protocol = tls - Enables TLS for this transport. SIP over TLS uses port 5061 by default.
• method = tlsv1_2 - Sets the minimum TLS protocol version. With OpenSSL 1.1.1+, both TLS 1.2 and TLS 1.3 are available.
• cipher - Restricts the allowed cipher suites to strong AEAD ciphers with forward secrecy.
• verify_server = yes - Verifies the server certificate when making outbound connections. Requires a valid ca_list_file.
• verify_client = no - Set to yes if you want to require client certificates for mutual TLS authentication.

See RFC 8446 §4.3.2 for the TLS Certificate Request specification, and Wikipedia: Mutual authentication for a general overview.

PJSIP Endpoint Configuration#

Configure endpoints to use the TLS transport:

[my-trunk]
type = endpoint
transport = transport-tls
context = from-trunk
disallow = all
allow = ulaw
allow = alaw
media_encryption = sdes
; ... other endpoint settings

The media_encryption = sdes setting enables SRTP for encrypting the audio stream in addition to the SIP signaling. Other options include:

• no - No media encryption
• sdes - SDES key exchange for SRTP (most common)
• dtls - DTLS-SRTP key exchange (more secure, but less widely supported)

SRTP for Media Encryption#

TLS only encrypts SIP signaling (call setup). To encrypt the actual audio/media, enable SRTP. With PJSIP, set media_encryption on the endpoint as shown above.

For mandatory SRTP (reject unencrypted media):

[my-endpoint]
type = endpoint
media_encryption = sdes
media_encryption_optimistic = no

Setting media_encryption_optimistic = no means the call will fail if SRTP cannot be negotiated, rather than falling back to unencrypted RTP.

Legacy chan_sip Configuration#

If you are still using the legacy chan_sip driver (not recommended for new deployments), configure TLS in sip.conf:

[general]
tlsenable = yes
tlsbindaddr = 0.0.0.0:5061
tlscertfile = /etc/asterisk/keys/asterisk.pem
tlsprivatekey = /etc/asterisk/keys/privkey.pem
tlscafile = /etc/asterisk/keys/ca.crt
tlscipher = ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305
tlsclientmethod = tlsv1_2

Note: tlsbindaddr = 0.0.0.0:5061 binds the IPv4 wildcard only. chan_sip accepts a single tlsbindaddr, so a single instance cannot bind both an IPv4 and an IPv6 wildcard for TLS; bind tlsbindaddr = [::]:5061 to serve IPv6 (which on most systems also accepts IPv4-mapped clients) or migrate to PJSIP for proper dual-stack with two transports.

Note: chan_sip was removed in Asterisk 21. It is not available in Asterisk 21 or later. New installations should use PJSIP.

Complete PJSIP Example#

A minimal pjsip.conf with TLS and SRTP:

; TLS transport
[transport-tls]
type = transport
protocol = tls
bind = 0.0.0.0:5061
cert_file = /etc/asterisk/keys/asterisk.pem
priv_key_file = /etc/asterisk/keys/privkey.pem
ca_list_file = /etc/asterisk/keys/ca.crt
method = tlsv1_2
cipher = ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305
verify_server = yes
verify_client = no

; IPv6 TLS transport (second transport for dual-stack)
[transport-tls-v6]
type = transport
protocol = tls
bind = [::]:5061
cert_file = /etc/asterisk/keys/asterisk.pem
priv_key_file = /etc/asterisk/keys/privkey.pem
ca_list_file = /etc/asterisk/keys/ca.crt
method = tlsv1_2
cipher = ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305
verify_server = yes
verify_client = no

; SIP trunk with TLS and SRTP
[my-provider]
type = endpoint
transport = transport-tls
context = from-external
disallow = all
allow = ulaw
allow = alaw
media_encryption = sdes
media_encryption_optimistic = no

[my-provider]
type = aor
contact = sip:sip.provider.example.com:5061\;transport=tls

[my-provider]
type = identify
endpoint = my-provider
match = sip.provider.example.com

Security Notes#

The cipher suite and protocol configuration in this guide addresses the following known TLS vulnerabilities:

• POODLE (CVE-2014-3566, 2014): SSL 3.0 is disabled. TLS_FALLBACK_SCSV was added in OpenSSL 1.0.1j / 1.0.2 (October 2014); SSL 3.0 disabled by default in OpenSSL 1.1.0 (August 2016).
• BEAST (CVE-2011-3389, 2011): Mitigated by recommending TLS 1.2 as the minimum; AEAD-only ciphers eliminate the CBC padding oracle.
• CRIME (CVE-2012-4929, 2012): TLS compression is off by default in OpenSSL 1.1.0+; do not enable it.
• Lucky13 (2013): AEAD-only cipher list eliminates CBC padding timing side-channels entirely.
• FREAK (CVE-2015-0204, 2015): EXPORT-grade ciphers are excluded. Removed from OpenSSL 1.1.0 (August 2016).
• LOGJAM (CVE-2015-4000, 2015): Short-key DHE is excluded; only ECDHE key exchange is recommended.
• Sweet32 (CVE-2016-2183, 2016): 3DES is excluded from the cipher string.
• ROBOT (2017): Static RSA key exchange is excluded; only ECDHE is recommended.
• Downgrade attacks: TLS_FALLBACK_SCSV prevents protocol version rollback.

The following are not addressable through TLS configuration alone:

• Heartbleed (CVE-2014-0160, 2014): A memory disclosure bug in OpenSSL 1.0.1 through 1.0.1f. Fixed in OpenSSL 1.0.1g (April 7, 2014). Addressed by patching OpenSSL, not by TLS configuration.
• BREACH (CVE-2013-3587, 2013): Not applicable. BREACH targets HTTP-level response compression; SIP/TLS signaling does not involve HTTP response compression.
• DROWN (CVE-2016-0800, 2016): Requires SSLv2 to be enabled on any server sharing the same private key. Ensure SSLv2 is disabled on all services that use the same certificate and key pair.

Verification#

Reload the PJSIP configuration and verify the TLS transport is active:

asterisk -rx "module reload res_pjsip.so"
asterisk -rx "pjsip show transports"

Verify TLS is working on the expected port:

openssl s_client -connect your-asterisk-server:5061

Check active channels for encryption:

asterisk -rx "pjsip show channels"