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I have a SIP provider: telecomitalia that use a cirpack!
The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I
don't know if it follows RFC...)
When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session
description is modified.
Both TRYING/RINGING with session description Connection Address/port are not replaced!
OpenWengo understand the modification and change the IP where it sends RTP from the one in
TRYING/RINGING SDP to the one in 200ok (proxy address
because of force_rtp_proxy())
But other softphones like Twinkle don't understand it and still send RTP to the first
IP!
Do you know how I can delete or modify TRYING/RINGING SDP?
Thanks
Below an example of a RINGING with SDP
#
U 212.129.6.65:5060 -> 88.191.45.91:5060
SIP/2.0 180 Ringing.
Allow: UPDATE,REFER.
Call-ID: 866489712(a)192.168.95.47.
Contact: <sip:212.129.6.65:5060>.
Content-Type: application/sdp.
CSeq: 21 INVITE.
From: "Marc LEURENT" <sip:mleurent@sd-7501.dedibox.fr>;tag=542924903.
Record-Route: <sip:88.191.45.91;lr;ftag=542924903>.
Server: Cirpack/v4.39a (gw_sip).
To: <sip:0614730696@sd-7501.dedibox.fr>;tag=01-07627-0003a50a-10152ba67.
Via: SIP/2.0/UDP
88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP
192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym.
Content-Length: 303.
.
v=0.
o=cp10 118890098142 118890098144 IN IP4 212.129.46.35.
s=SIP Call.
c=IN IP4 212.129.47.194.
t=0 0.
m=audio 32896 RTP/AVP 8.
b=AS:64.
a=rtpmap:8 PCMA/8000/1.
a=ptime:10.
a=sendrecv.
m=video 65534 RTP/AVP 34 31.
a=rtpmap:34 H263/90000/1.
a=fmtp:34 .
a=rtpmap:31 H261/90000/1.
a=fmtp:31 .
a=inactive.
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