Port 5061: SIPS — The Encrypted Voice of the Internet

Port 5061 carries SIPS traffic: the Session Initiation Protocol secured with Transport Layer Security. Every time a VoIP phone establishes an encrypted call, every time your office phone system authenticates without exposing your credentials to the network, every time a video conference begins with the assurance that the call metadata itself is protected, port 5061 makes that possible.

If port 5060 is the open conversation in a crowded room, port 5061 is the same conversation behind a locked door.

What Port 5061 Does

SIP (Session Initiation Protocol) is the signaling protocol that establishes, modifies, and terminates voice and video calls over IP networks.¹ It does not carry the actual audio or video. That is the job of RTP (Real-time Transport Protocol). SIP handles the handshake: "I want to call this person. Here are my credentials. Here is where to send the media. The call is over now."

On port 5060, all of this happens in plaintext. Your username. Your password. The phone number you are calling. The IP addresses involved. Anyone with access to a network tap can read it all.²

Port 5061 wraps that entire conversation in TLS encryption.³ The protocol is the same. The messages are the same. But now they are encrypted before transmission, and decrypted only by the intended recipient. The SIPS URI scheme (sips:user@example.com) explicitly demands this protection.⁴

How SIPS Works

The mechanism is straightforward:

1. A client initiates a TCP connection to port 5061 on the SIP server
2. TLS handshake begins: the server presents its certificate, encryption parameters are negotiated
3. Once the secure channel is established, SIP messages flow through it
4. The server (or next hop) decrypts the messages, processes them, and potentially re-encrypts for the next segment

This is hop-by-hop security, not end-to-end. Each SIP proxy in the path terminates TLS and may establish a new TLS connection to the next hop. The message content is visible to each proxy it passes through. This is by design: SIP proxies need to read and sometimes modify headers to route calls correctly.⁵

For true end-to-end protection of the media streams (the actual voice and video), you need SRTP (Secure Real-time Transport Protocol), defined in RFC 3711.⁶ SIPS protects the signaling. SRTP protects the conversation itself. Both are needed for comprehensive VoIP security.

The History

SIP was created in 1996 by Mark Handley, Henning Schulzrinne, Eve Schooler, and Jonathan Rosenberg.⁷ They were not trying to reinvent the telephone. They were trying to solve a problem with multicast multimedia on the Mbone.

The Mbone (Multicast Backbone) was an experimental overlay network built on top of the Internet in 1992.⁸ It enabled one-to-many streaming: a single source could broadcast audio and video to thousands of receivers without sending thousands of separate streams. IETF meetings were audiocast over it. In 1993, the first movie was streamed over the Internet using Mbone.⁹ In 1994, the Rolling Stones performed the first major multicast concert.

But coordinating these sessions was a mess. How do you announce that a session exists? How do participants join? How do you manage who is speaking? The researchers needed a signaling protocol.

The protocol they designed was text-based, modeled on HTTP and SMTP.¹ This was a deliberate choice. It made debugging possible (you could read the messages), interoperability easier, and the learning curve gentler. It also meant that anyone who could read network traffic could read your SIP messages.

SIP was first standardized as RFC 2543 in 1999, then substantially revised as RFC 3261 in June 2002.¹ By then, VoIP was real. Enterprises were replacing PBX systems with IP-based alternatives. The protocol designed for academic multicast sessions was now carrying business communications.

RFC 3261 established that port 5060 would be the default for unencrypted SIP, and port 5061 would be the default for SIP over TLS.³ The choice of adjacent port numbers was no accident: it signaled that these were the same protocol, just with different security properties.

Jonathan Rosenberg, the primary author of RFC 3261, was included in MIT Technology Review's TR35 list in 2002 as one of the world's top innovators under 35.¹⁰ Network World called him "a pioneer [in] the development of the SIP protocol." The protocol he helped create now underpins virtually all modern VoIP communication.

The Security Problem

Unencrypted SIP is a security disaster waiting to happen.

The attack surface includes:¹¹

Eavesdropping: SIP and RTP traffic in cleartext can be captured and replayed. Tools like VOMIT (Voice Over Misconfigured Internet Telephones) and SIPTap demonstrated this years ago. An attacker on the network can listen to your calls in real time.

Credential theft: SIP REGISTER messages contain authentication information. Without TLS, these credentials travel in the clear. Attackers can capture them and impersonate legitimate users.

Call hijacking: Man-in-the-middle attacks can redirect SIP sessions to rogue endpoints. Academic studies have shown that attackers can modify SIP messages in transit without detection, including the SDP (Session Description Protocol) contents that specify where media should be sent.

Toll fraud: Attackers gain access to SIP accounts and make expensive international calls. SIPvicious is an open-source tool that scans networks for SIP servers and exploits weak credentials. The resulting fraud costs enterprises millions annually.

Denial of service: SIP infrastructure can be overwhelmed with bogus requests, disrupting communications for legitimate users.

Port 5061 and TLS encryption address many of these attacks. Eavesdropping becomes computationally infeasible. Credential theft requires compromising the TLS connection itself. Message modification triggers TLS integrity checks. Ghost calls and scanning attempts can be blocked at the TLS handshake stage before any SIP processing occurs.

But TLS is not magic. It protects the path, not the endpoints. A compromised SIP proxy can still see all traffic passing through it. And the SIPS scheme guarantees TLS only to the domain boundary. What happens inside the destination network depends on local policy.⁴

RFC 5630, published in 2009, clarified the meaning and limitations of the SIPS URI scheme.⁴ It explicitly warned against treating SIPS like HTTPS: "Some have been tempted to believe that the SIPS scheme was equivalent to an HTTPS scheme in the sense that one could provide a visual indication to a user (e.g., a padlock icon) to the effect that the session is secured. This is obviously not the case."

Honesty about these limitations is essential. SIPS provides meaningful protection against network-level attacks. It does not provide end-to-end security guarantees.

The Modern Landscape

SIP and SIPS are everywhere.

The global VoIP market was valued at $144.77 billion in 2024 and is projected to reach $326.27 billion by 2032.¹² SIP trunking, which replaces traditional phone lines with SIP-based connections, held a 32.2% market share in 2024. More than 60% of companies globally use VoIP for business communications. In the United States, approximately 78% of small businesses use VoIP phone systems.

The shift to remote and hybrid work has accelerated adoption. SIP trunking allows employees to use their business phone numbers from anywhere with an Internet connection. The cost savings are substantial: switching from traditional phone lines to SIP can reduce communication costs by 20% to 60%.

Increasingly, that traffic flows over port 5061. Compliance requirements (HIPAA in healthcare, PCI-DSS for payment processing, GDPR in Europe) often mandate encryption of voice traffic. Enterprises handling sensitive information cannot afford to transmit call metadata in cleartext. Port 5061 is no longer optional for many organizations.

The protocol that started on the Mbone now carries billions of calls. The security that was an afterthought in the 1990s is now a regulatory requirement. And the port that was reserved for "SIP over TLS" has become essential infrastructure.

Related Ports

Port 5060 and 5061 are siblings: same protocol, different security posture. RTP and RTCP handle the actual media that SIP establishes. For secure media, SRTP (typically on dynamic ports negotiated via SIP/SDP) encrypts the audio and video streams.