Hello,
you can try delete and rebuild Contact Header before send it to FreeSWITCH.
Regards
El 4/02/2025 a las 2:35 p. m., James Lipski via sr-users escribió:
Hello,
I have a kamailio installation where it's basically doing transport
protocol translation. To go into more detail, I have a UA
communicating over TLS towards kamailio/ and kamailio forwards the
request to a freeswitch installation via UDP, so call-flow is as follows:
UA <-- (TLS) --> Kamailio <-- (UDP) --> Freeswitch
For the most part, this does work/ it's only when I use a UA that uses
"sips:" as the schema instead of "sip:" where issues occur -- in my
current topology, freeswitch has some weird behavior where it doesn't
disconnect the call properly. At least based on my testing, I believe
the issue is relating to the schema -- as the UA in question does have
the ability to switch between schemas/ when using "sip:" instead of
"sips:" over TLS, no issues with the same call-flow. In the event that
I do come across a UA that doesn't have that ability, I was wondering
if kamailio has the ability to switch schemas when it sends traffic
over the UDP connection.
I'm able to replace most aspects of the packet with "sip:" however the
only header which contains the "sips:" schema is the contact header.
With this alone, it propagates to the B-leg side (UDP side) of the
call; and will cause freeswitch to not disconnect calls properly.
I'm using the topos module to obfuscate my topology, not sure if that
matters.
--------
For example, on the A-leg side of the call, kamailio receives this:
INVITE sips:[email protected]:5061 SIP/2.0
Via: SIP/2.0/TLS 172.16.1.194:5061;branch=z9hG4bK236557042;rport;alias
From: <sips:[email protected]>;tag=444048970
To: <sips:[email protected]>
Call-ID: [email protected]
CSeq: 21 INVITE
Contact: <sips:[email protected]:5061;transport=tls>
Proxy-Authorization: Digest username="100000000",
realm="dev-sip.server.com <http://dev-sip.server.com>",
nonce="Z6Ig0WeiH6Vdxtvej0I3XGCTUAEfh+0/",
uri="sips:[email protected] <mailto:uri=>",
response="43c1487d077db36a70fffdd1e25b3b52", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT812 1.0.57.1
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 1034
v=0
o=100000000 8000 8000 IN IP4 172.16.1.194
s=SIP Call
c=IN IP4 172.16.1.194
t=0 0
m=audio 5004 RTP/SAVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=crypto:1 AEAD_AES_256_GCM
inline:UlX3PAjNBP/aQwtwyf5ymsfJciB+VdFxMElO5zLVooUqmcEzZ8YyQQk9stI=|2^32
------------------------------------------------------
On the B-leg side of the call, kamailio forwards the following to
freeswitch:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
10.64.52.113:5080;branch=z9hG4bK1552.89ec9666294034a5611f908f2a6a427e.0;i=f
From: "Test" <sip:[email protected]>;tag=444048970
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 21 INVITE
Max-Forwards: 15
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 316
Contact: <sips:[email protected]:5080>
v=0
o=100000000 8000 8000 IN IP4 10.64.52.113
s=SIP Call
c=IN IP4 10.64.52.113
t=0 0
m=audio 13060 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
--------------------------------------------------
As with the above, I was wondering if there is a way within kamailio
to switch schemas on the b-leg side of this call example; specifically
is there a way to update the contact header so it doesn't use the
"sips:" schema when it traverses the UDP connection.
Thank you for you time.
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