Re: topoh breaks SIPS with PJSIP if PJMEDIA_SRTP_MANDATORY is set

Володимир Іванець via sr-users <[email protected]>
Newsgroups gmane.comp.voip.ser
Message-ID <CAOQgkjbEW7HxyqJSw6qYoOnw=fnWi10Mo+hbFGU5TbJ5+AmHCQ@mail.gmail.com>
Hello!

I just faced the same behavior.

I have modparam("topoh", "mask_ip", "") so Kamailio uses the address it
sends packets from. Inside event_route[topoh:msg-outgoing] it rewrites the
Contact header but skips adding transport value for TCP or TLS. Is there a
way of including transport value or adding parameter to mask_mode setting
to exclure processing Contact header for all responses?

Thanks!

пт, 22 серп. 2025 р. о 17:09 Mathias Schneuwly via sr-users <
[email protected]> пише:

> Hi
>
> I have a question related to topoh. We are running Kamailio in version
> 6.0.2 with topoh enabled. The PJSIP based softphone is unable to establish
> a call with SIPS and SRTP if we use "PJMEDIA_SRTP_MANDATORY" in PJSIP.
>
> PJSIP complains about "Secure dialog requires SIPS scheme in Contact and
> Record-Route headers, ending the session". The other side is sending
> "Contact: <sips:[email protected]:59686;transport=tls>", but topoh
> modifies the contact header to "Contact:
> <sip:127.0.0.8;line=sr-avdFaQDvhUxGrqtxhqxThq-ZnUy6nUzGrQDZmqa1rUf6a0JTaLt4aoyV2EGQ>"
> without sips scheme and without transport=tls. PJSIP then refuses to
> establish the call and sends immediately a BYE.
>
> If topoh is disabled, the call is successfully established.
>
> Is it somehow possible to tell topoh to keep the scheme or transport=tls
> if the other side provided it?
>
> Thanks for your help
> Best regards
> Mathias
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