3.1.5.1 Overview of Authentication Protocol Elements. 2/14/2019; 4 minutes to read; In this article. The server issues an authentication challenge by using either a 401 or 407 response to a SIP request. The client SHOULD be capable of processing either response. The server uses the following SIP headers as part of this authentication scheme. SIP v5 is the current stable release. It will be supported until the release of SIP v6. This is a snapshot of the next stable release including all bug fixes. This is the first release without the legacy API. If you have not migrated, 0.15.x contains both the legacy and new APIs. The new API on 0.15.11 is the same as the only API on this version.
SIP v5.4.0 Released. SIP v5.4.0 has been released. This is a feature release including support for Python v3.9. This is the last release that will support Python v3.5. The latest version of the module ABI is v12.8.1. Python v3.9 is supported. Added support for building the sip module for PyPy. SIP/2.0 183 Session in Progress Via: SIP/2.0/UDP 10.135.0.12:5060;branch=z9hG4bKw5ij3wn4knq9hn8kkrmoa64.ahrre7d;rport= 5060 From: 'Calling User' sip:[email protected]:5060;tag=m3l2hbp To: sip:[email protected]:5060;user=phone Call-ID: [email protected] CSeq: 10692 INVITE Server: Wildix GW-4.2.5.35963 Content-Type: application/sdp.
This is a VoIP module for the honeypot dionaea. The VoIP protocol usedis SIP since it is the de facto standard for VoIP today. In contrast tosome other VoIP honeypots, this module doesn’t connect to an externalVoIP registrar/server. It simply waits for incoming SIP messages (e.g.OPTIONS or even INVITE), logs all data as honeypot incidents and/orbinary data dumps (RTP traffic), and reacts accordingly, for instance bycreating a SIP session including an RTP audio channel. As sophisticatedexploits within the SIP payload are not very common yet, the honeypotmodule doesn’t pass any code to dionaea’s code emulation engine. Thiswill be implemented if we spot such malicious messages. The mainfeatures of the VoIP module are:
- Support for most SIP requests (OPTIONS, INVITE, ACK, CANCEL, BYE)
- Support for multiple SIP sessions and RTP audio streams
- Record all RTP data (optional)
- Set custom SIP username and secret (password)
- Set custom useragent to mimic different phone models
- Uses dionaea’s incident system to log to SQL database
Personalities¶
A personality defines how to handle a request. At least the ‘default’personality MUST exist. The following options are available perpersonality.
serve
A list of IP addresses to use this personality for.
handle
SIP Users¶
You can easily add, change or remove users by editing the SQLite filespecified by the ‘users = “”’ parameter in the config file. All usersare specified in the users table.
username
Specifies the name of the user. This value is treated as regularexpression. See Python: Regular Expressions<http://docs.python.org/py3k/library/re.html> for more information.
password
personality
The user is only available in the personality specified by thisvalue. You can define a personality in the config file.
pickup_delay_min Djay pro 2 2 0 7.
This is an integer value. Let the phone ring for at least thisnumber of seconds.
pickup_delay_max
This is an integer value. Maximum number of seconds to wait beforedionaea picks up the phone.
action
sdp
The name of the SDP to use. See table ‘sdp’.
SDP¶
All SDPs can be defined in the sdp table in the users database.
name
sdp
The value to use as SDP
![Sip 102 Sip 102](https://www.researchgate.net/profile/Johnson_Agbinya/publication/4167656/figure/tbl1/AS:669007283773442@1536515174205/Table4-shows-the-disruption-time-as-the-delay-DmTOc-increases-Here-the-DoTOn-are-assumed.png)
The following values are available in the SDP definition.
{addrtype}
{unicast_address}
RTP address
Sip 1 0 8 0
{audio_port}
{video_port}
Dionaea video port.
The following control parameters are available in the SDP definition.
[audio_port]…content…[/audio_port]
The content is only available in the output if the audio_port valueis set.
[video_port]…content…[/video_port]
The content is only available in the output if the video_port valueis set.
Example:
Example config¶
-->Specifies Microsoft extensions to the Session InitiationProtocol (SIP), as specified in [RFC3261], which is used by terminals toestablish, modify, and terminate multimedia sessions or calls. The SIPextensions add support for privacy features and for subscription requests foroffline end nodes to the SIP extensions for presence.
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Published Version
Date | Protocol Revision | Revision Class | Downloads |
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6/1/2017 | 8.0 | None | PDF | DOCX | Diff |
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How many daw users worldwide. 8/14/2009 | 5.0.4 | Editorial | |
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6/20/2008 | 2.0 | Major | |
5/16/2008 | 1.0.3 | Editorial | |
3/14/2008 | 1.0.2 | Editorial | |
1/25/2008 | 1.0.1 Kannada audio songs. | Editorial | |
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8/10/2007 | 0.2 | Minor | |
5/11/2007 | 0.1 | New |
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Sip 102 Bye
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