[asterisk-bugs] [JIRA] (ASTERISK-24428) Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24428?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team updated ASTERISK-24428:
-------------------------------------

    Target Release Version/s: 16.10.0

> Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used
> ----------------------------------------------------------------------------------------------------------------------------
>
>                 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: SVN, 1.8.32.0, 11.12.0, 12.7.0, 13.0.0
>         Environment: Ubuntu 14.04, OpenWrt 14.07
>            Reporter: sstream
>      Target Release: 13.33.0, 16.10.0
>
>
> 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 and 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 used.
> 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. bindport=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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