Do you have set_contact_alias or add_contact_alias in Kamailio? Anyways you're doing something wrong as AFAIK Kamailio translates contact header to udp automatically. You should try to post on sr-users list.

2016-11-22 12:33 GMT+02:00 Vladyslav Zakhozhai <v.zakhozhai@gmail.com>:
Hi,

I'm trying to understand what is the best or suitable approach to the following use case. Let me simplify thing a little bit. 

Suppose we have one FreeSWITCH registrar behind SIP proxy (kamailio). I'd like to offload SSL/TLS encryption/decryption to SIP proxy:

REGISTER:

Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50
Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH

INVITE:
UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> Kamailio == SIP/TLS ==> UAC2

(FreeSWITCH uses kamailio as outbound proxy with fs_path tag appended in dialplan).

The main problem is in Contact header which contains transport=tls and we can see it in FreeSWITCH console:

User:       user@domain.com
Contact:   "" <sip:user@UAC_IP:57976;transport=tls>
Status:     Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) EXPSECS(108)
IP:         SIP_PROXY_IP
Port:       5060

When FreeSWITCH sends INVITE to UAC2 (during call) it tries to establish TLS session to UAC2. It fails because there is no TLS-enabled sofia profiles in the config of FreeSWITCH.

I have only one solution in my mind: rewrite transport tag in Contact header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls to UAC).

I'd like to know it this solution ok or there is more elegant solutions.

I've tried appending tag transport=udp in FreeSWITCH's dialplan but no success.

Thank you in advance.

--
С уважением,
Владислав Захожай


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--
Alexandru Covalschi
VoIP engineer and system administrator
tel: +37367398493