Send the History-Info header, conveying the diversion information to the called and calling user agents. Determines whether encryption should be used if possible but does not terminate the session if not achieved. Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side. This is a comma-delimited list of auth sections defined in pjsip.

rtcp-mux in WebRTC

Endpoints without an authentication object configured will allow connections without verification. Using the same auth section for inbound and outbound authentication is not recommended. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses. See the auth realm description for details. Endpoints and AORs can be identified in multiple ways. This option is a comma separated list of methods the endpoint can be identified. This option controls both how an endpoint is matched for incoming traffic and also how an AOR is determined if a registration occurs.

You must list at least one method that also matches for AORs or the registration will fail. This method of identification has some security considerations because an Authentication header is not present on the first message of a dialog when digest authentication is used. The client can't generate it until the server sends the challenge in a response. Since Asterisk normally sends a security event when an incoming request can't be matched to an endpoint, using this method requires that the security event be deferred until a request is received with the Authentication header and only generated if the username doesn't result in a match.

This may result in a delay before an attack is recognized. When a redirect is received from an endpoint there are multiple ways it can be handled.

If this option is set to user the user portion of the redirect target is treated as an extension within the dialplan and dialed using a Local channel. More than one mailbox can be specified with a comma-delimited string. On inbound SIP messages from this endpoint, the Contact header or an appropriate Record-Route header will be changed to have the source IP address and port.

This option does not affect outbound messages sent to this endpoint. This option helps servers communicate with endpoints that are behind NATs. When enabled, immediately send Ringing or Progress response messages to the caller if the connected line information is updated before the call is answered. This can send a Ringing response before the call has even reached the far end. The caller can start hearing ringback before the far end even gets the call.

Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of this Asterisk box. When disabled, a connected line update must wait for another reason to send a message with the connected line information to the caller before the call is answered. You can trigger the sending of the information by using an appropriate dialplan application such as Ringing.

This will force the endpoint to use the specified transport configuration to send SIP messages. Not specifying a transport will select the first configured transport in pjsip.We use optional third-party analytics cookies to understand how you use GitHub. Learn more. You can always update your selection by clicking Cookie Preferences at the bottom of the page.

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Analytics cookies We use analytics cookies to understand how you use our websites so we can make them better, e. Save preferences. PBX Core settings. Version: Did you notice calls stop working after updating Google Chrome to version 57?

Are you curious why that happened? The answer is the rtcp-mux feature.

asterisk disable rtcp

Extensions to RTCP also allow for it to be used for protocol-specific controls to the stream, such as indicating to a sender to send a full frame of video. In other words, demultiplexing of the traffic types is performed at the transport layer.

Simply by knowing the port that traffic is coming in on, you know what type of packet you are receiving. This offers some advantages over the old method:. Google Chrome has had rtcp-mux capabilities for several years. In addition, Chrome and Asterisk guru Dan Jenkins wrote a blog post that details the Chrome changes, how they affect Asterisk, and how you can work around them if necessary.

This means that when placing calls to Asterisk, Chrome would fall back to using traditional RTCP since Asterisk did not support rtcp-mux. Their reasoning for doing so was two-fold:. To get around this problem, the Asterisk team decided to add support for rtcp-mux into Asterisk before it became too late. The feature is available starting in Asterisk For those of you still on older versions, you may want to start upgrading soon if you plan to interoperate with Chrome.

Your email address will not be published. Skip to content. Mark Michelson. What is rtcp-mux? This offers some advantages over the old method: It simplifies NAT traversal since only a single port is used for media and control messages.

You can theoretically double the amount of media sessions on your system with the same number of UDP ports.

VoIP Traffic Analysis: SIP + RTP

You only need one set of candidates instead of two. Their reasoning for doing so was two-fold: rtcp-mux is used by the vast majority of their WebRTC traffic. With this switchover, calls from Chrome to Asterisk started failing.

Leave a Comment Cancel Reply Your email address will not be published. About the Author. See All of Author's Posts. What can we help you find? Download Asterisk. Get Started. Asterisk Applications Features Glossary. Other Resources. Docs Blog Forums Training Join.This month, the Asterisk project performed two security releases to address an unauthorized RTP data disclosure vulnerability in its real-time transport protocol RTP stack.

If a malicious actor knew the RTP ports for a session, or simultaneously sent packets to all potential RTP ports, and could send enough RTP packets in an established stream, then Asterisk would lock onto the malicious actor as the RTP source. This potentially allowed a malicious entity to temporarily deny, redirect, or capture an RTP stream in an established call, or perform a distributed denial of service DDOS attack on the system. While security vulnerabilities can happen in any software project, the fact that this particular vulnerability resulted in an incomplete fix that necessitated a second security release has led us to want to describe in detail what occurred.

In this blog post we are going to explore two different topics. Before we dig into the specifics of the vulnerability, it is important to understand some of the issues that surround RTP as a protocol, as well as the various mechanisms that exist in Asterisk to work with and around those limitations.

While this list is not exhaustive, the following three combined create the situation which led to the vulnerability in Asterisk. This includes:. Without this information, an attacker would have to continually send a large number of packets over a large port range in order to disrupt RTP media streams, which is often infeasible but certainly possible.

Thus, if your endpoint supports communicating with Asterisk via TLS, all of your SIP message traffic and the SDP message bodies can be encrypted, and malicious attackers cannot intercept the messages to gain information about the resulting RTP session.

While SIP messages can be authenticated, the RTP protocol by itself — with no help from other protocols — does not define any authentication provisions. Without involving some other protocols, there is no foolproof way to determine if the RTP packet you received is from a valid sender. Since version 1. With the media stream encrypted, it is extremely difficult if not impossible for a malicious actor to affect the RTP stream. Asterisk provides mechanisms that should always be used to help prevent unauthorized RTP traffic from being processed within a session:.

It has no routable path to that IP address, and it may even have multiple phones communicating with it that are advertising that same IP address! While some NAT devices may implement a SIP ALG to assist with advertising the correct IP address and port and, in some cases, to make things much worseand while there are some protocols that endpoints may use to help discover their publicly accessible IP address and port, often these things are not available or sufficient for media transmission.

Typically, NAT devices will relay packets through that port back to the endpoint that sent the media. Most endpoints, if they receive an RTP packet on the same port that they transmitted RTP from, will recognize that they are behind a NAT and process the media as if it were received on the advertised port.

This allows endpoints to punch a hole through the NAT, establishing bidirectional media without the assistance of an ALG or other protocols. For a reason we will get into in a little bit, Asterisk had been changed in Asterisk In all scenarios, the non-hijacked leg of the call in Asterisk will hear garbled audio. Why did it break? As is so often the case with security vulnerabilities, the vulnerability was introduced as a result of a fix to a bug found in testing.

The scenario we were attempting to address was, admittedly, rather challenging:. When a call is established from Phone 1 through both Asterisk servers to Phone 2, the resulting media flow should look like what is shown in the diagram below:. While this is happening, however, Asterisk will forward RTP packets between both endpoints, so that there is minimal interruption in media.GitHub is home to over 50 million developers working together to host and review code, manage projects, and build software together.

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asterisk disable rtcp

We use essential cookies to perform essential website functions, e. We use analytics cookies to understand how you use our websites so we can make them better, e. Skip to content. Code Pull requests Actions Security Insights. Permalink Dismiss Join GitHub today GitHub is home to over 50 million developers working together to host and review code, manage projects, and build software together.

Sign up. Go to file T Go to line L Copy path. Raw Blame. Strict RTP qualifies RTP ; packet stream sources before accepting them upon initial connection and ; when the connection is renegotiated e. Once Asterisk has recognized a stream it will ; allow other streams to qualify and replace the current stream for 5 ; seconds after starting learning mode. Once learning mode completes the ; current stream is locked in and cannot change until the next ; renegotiation.

Consider changing this value ; if rtp packets are dropped from one or both ends after a call is ; connected. This option is set to 4 by default. This option is enabled by default. The port number is ; optional.

If omitted the default value of will be used. This option is ; disabled by default. Blacklisting is done via ACL infrastructure ; so it's possible to whitelist as well.This page contains a list of necessary tasks in order to write a new RTP engine.

The tasks start out with incredibly specific detail and eventually move into much more general terms. The reason for this is to help remain agile in the process. Undoubtedly, requirements will change, or mistakes in the design early will change the nature of later tasks. The first milestone will be to get the RTP engine written to the point that we can successfully pass media through Asterisk.

This means successfully constructing the media flows as described in the parent page and putting some basic setup in place. For this phase, it is encouraged to use hard-coding in place of pluggable elements.

The sooner we can get to a point where we have successful calls, the easier it is to rapidly develop new features and tests! The goal of this task is basically to make sure that when channel drivers attempt to interact with the RTP engine, they do not crash. Fill in all required functions with stubs that return some error value.

All optional functions should be left empty for now. Required functions are:. The RTP stream structure will have nothing on it for the time being. We will add the requisite parts to it as tasks require them.

We will add parts to it as tasks require them. The first task is to be able to play audio over an RTP stream.

In order to reach this, the media outflow will need to be implemented. According to the planned outflow, a media write should go through an encoder, a router, and a transport. Create an RTP encoder structure. Since this is the bare-bones phase, the RTP encoder can cheat in some respects, such as always using the same SSRC and never setting the marker bit.

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However, it must properly indicate sequence numbers and timestamps. Create a default packet router structure. Add this structure to the RTP stream structure. The main job of the default packet router is to provide an address for the packet to be routed to.

In order to actually transmit media, the RTP engine's write function will need to be implemented. For this task, the write function will operate like so. The next logical task is to be able to create the media inflow. With this task completed, a call should be capable of receiving media and passing it up to the channel when requested. Once we have the bare minimum setup created and tested, we can start to add more basic RTP features.

For this particular task, though, we're going to leave this method stupid. The marker bit should always be set during the first packet that we send during an RTP stream. Configuration is a necessary predecessor for getting rid of some hard-coded values in the engine.

Asterisk: Console Commands

If binding fails, the number is incremented by 2 and binding is retried. It is unknown how much this impedes RTP stream setup times on loaded systems, so you will want to implement port selection as a function that can be replaced with a new algorithm if desired.

One naive suggestion for a different port selection algorithm would be to have all legal ports defined by the configured range in a queue with the order randomized.

asterisk disable rtcp

Pop a number off the queue when it is time to allocate a port. When you have finished using that port, push it back onto the queue.Let our experts guide you to victory with their best football tips selection of the day.

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