Menu Close

Tag: sip

Using tcpdump for SIP diagnostics

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:

SIP clients – security features analysis

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

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

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.