[asterisk-bugs] [JIRA] (ASTERISK-24428) Can't disconnect Asterisk server using non-standard TLS port (other than 5061)

[sstream (JIRA)]

sstream created ASTERISK-24428:
----------------------------------

             Summary: Can't disconnect Asterisk server using non-standard TLS port (other than 5061)
                 Key: ASTERISK-24428
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24428
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_sip/TCP-TLS
    Affects Versions: 11.12.0
         Environment: Ubuntu 14.04, OpenWrt 14.07
            Reporter: sstream


Using Asterisk 11.12 server behind NAT (router) and Android SIP client (Acrobits or CSipSimple) with the following condition.
- Router local IP: 192.168.0.1
- Router domain: sip.example.net (using DNS like DynDNS)
- Asterisk server local IP: 192.168.0.2
- SIP protocol: TLS, port=10000 (not default due to security reason)
- RTP: port=10001 to 10008
- Android phone has a global IP.

In router's setting, I opened ports 10000-10008 (TLS and RTP) and forwarded them to Asterisk server IP (192.168.0.2).

--sip.conf--
externhost=sip.example.net
localnet=192.168.0.0/255.255.255.0

tlsenable=yes
tlsbindaddr=0.0.0.0:10000
tlscertfile=…
---
--rtp.conf--
rtpstart=10001
rtpend=10008
---

Under above condition, there is no problem for registration, connection, calling, talking.
However, Asterisk cannot detect disconnection. (can't disconnect).

I traced a log and found;
---
SIP/2.0 100 Trying
Via: SIP/2.0/TLS xxx.xxx.xx.0:xxxxx;branch=;received=;rport=
From: <sips:xxx@ sip.example.net:10000>;tag=xxx
To: <sips:yyy@ sip.example.net:10000>
Call-ID: xxxxx
CSeq: 1 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:yyy at x.xx.xx.xx:5061;transport=TLS>
Content-Length: 0
---

In "Contact:" line (2nd line from the bottom), port “5061” suddenly appears, which I never use.

This problem only occurs under TLS with non-default port (not 5061).
There is no problem under UDP, even though  non-standard port is set (ex. bindaddr=10000).

For testing, in “sip.h”, I changed
#define STANDARD_TLS_PORT	5061
to
#define STANDARD_TLS_PORT	10000
and compiled.
As a result, I confirmed that Asterisk works properly (detect disconnection).

In conclusion, it is better to make an option like “tlsbindport” in sip.conf and make it work like UDP.

Thank you.




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