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Category: SIP

Tools for a quick SIP diagnostics – ngrep, sipgrep and sngrep

palo73 0 Comments

Sometimes there is a need for simple and quick analysis or the troubleshooting of a SIP server and its call functions. Of course, we should use the well-known tcpdump, mentioned in the article Using tcpdump for SIP diagnostics. However, for some occasional Linux users this may be too difficult and unclear. Actually there exist some simpler utilities,, that could work fine, as ngrep, and for me newer sipgrep and sngrep (love at first sight).

All utils are directly available and can be installed online from Debian repo using apt-get install ngrep sipgrep sngrep.

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Using tcpdump for SIP diagnostics

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TCPdump is a powerful command-line packet analyzer, which may be used for a SIP message sniffing/analyzing. TCPdump is preinstalled on many linux distributions, or may be installed directly from debian repository:

apt-get install tcpdump

TCPdump allows write sniff to a file or display it realtime. Its usage for SIP message analysis may look like:

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Installing and configuring Homer SIP capture server – debian 64b squeeze – howto

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Description: Homer is an Open Source SIP Capture server by Alexandr Dubovikov & Friends, based on OpenSER/Kamailio and supporting HEPv1/v2 (Homer Encapsulation Protocol) & IP proto 4 (IPIP) encapsulation and monitoring/mirroring port capture modes. Homer ships with a flexible and lightweight capture agent for unsupported scenarios and a powerful browser based UI (webHomer).

Web:

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SIP clients – security features analysis

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Table provides the overview of security features of nine analysed open-source SIP clients (some sources call them the RTC communicator).

Source: P. Segeč, M. Moravčík, J. Hrabovský, J. Papán and J. Uramová, “Securing SIP infrastructures with PKI — The analysis,” 2017 15th International Conference on Emerging eLearning Technologies and Applications (ICETA), Stary Smokovec, 2017, pp. 1-8.
doi: 10.1109/ICETA.2017.8102525
URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?rp=&arnumber=8102525&isnumber=8102457

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Problem with a VoIP phone behind NAT – disabling FortiGate SIP ALG

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Initial state and observed problems

Observed problems

We had observed a problem, where a SIP phone is registering, but the AOR record indicates, that as a Contact IP address the incorrect and strange private IP address is used. As is shown on following listing:

voip*CLI> pjsip show aor 1765
   Aor:  <Aor..............................................>  <MaxContact>
     Contact:     
   Aor:  1765                                                 1 Contact:  1765/sip:1765@10.16.42.46:65476              f123d14d1c NonQual         nan
 ParameterName        : ParameterValue
  =================================================
  authenticate_qualify : false
  contact              : sip:1765@10.16.42.46:65476
  default_expiration   : 7200
  mailboxes            :
  max_contacts         : 1
  maximum_expiration   : 7200
  minimum_expiration   : 60
  outbound_proxy       :
  qualify_frequency    : 0
  qualify_timeout      : 3.000000
  remove_existing      : true
  support_path         : false
  voicemail_extension  :

This cause a problem, where incoming phone calls (call on 1765 number) are not reaching the SIP phone. We had tried to solve the situations on the phone only modifying its NAT configuration and using STUN, but with no success. Then we setup the lab with two Cisco NAT to simulate the topo. It works perfectly. This indicate on a problem with the Fortigate firewall. Several posts indicates that it could be the SIP ALG problem, which is on Fortigate devices turned on by default and it modifies SIP messages.

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