4 Sep
2007
4 Sep
'07
1:46 p.m.
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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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4 Sep
4 Sep
1:49 p.m.
Hi Marc,
this one should be handled in the same way u do with 183/200, catch it
in onreply route and force_rtpproxy()
Cheers,
DanB
On 9/4/07, Marc LEURENT <lftsy(a)free.fr> wrote:
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Hash: SHA1
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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_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
2 p.m.
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Hash: SHA1
I already have:
onreply_route[1] {
xlog("!!!! STARTING REPLY ROUTE\r\n");
#uac_restore_from();
if ((isflagset(5) || isbflagset(6)) &&
status=~"(183)|(2[0-9][0-9])") {
force_rtp_proxy();
}
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) {
fix_nated_contact();
}
exit;
}
???? Strange ?? no?
Dan-Cristian Bogos a écrit :
Hi Marc,
this one should be handled in the same way u do with 183/200, catch it
in onreply route and force_rtpproxy()
Cheers,
DanB
On 9/4/07, Marc LEURENT <lftsy(a)free.fr> wrote:
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.
>
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
>
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2:41 p.m.
Marc,
Not at all strange. As depicted by you also, you only have it set for
183 and 200. You need a line like this:
onreply_route[1] {
if ((isflagset(5) || isbflagset(6)) &&
status=~"(100|180|183)|(2[0-9][0-9])") {
.........
}
Cheers,
DanB
On 9/4/07, Marc LEURENT <lftsy(a)free.fr> wrote:
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Hash: SHA1
I already have:
onreply_route[1] {
xlog("!!!! STARTING REPLY ROUTE\r\n");
#uac_restore_from();
if ((isflagset(5) || isbflagset(6)) &&
status=~"(183)|(2[0-9][0-9])") {
force_rtp_proxy();
}
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) {
fix_nated_contact();
}
exit;
}
???? Strange ?? no?
Dan-Cristian Bogos a écrit :
Hi Marc,
this one should be handled in the same way u do with 183/200, catch it
in onreply route and force_rtpproxy()
Cheers,
DanB
On 9/4/07, Marc LEURENT <lftsy(a)free.fr> wrote:
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.
>
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
>
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3 p.m.
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
One day I will learn to think twice before saying bullshits!
Thanks
Dan-Cristian Bogos a écrit :
Marc,
Not at all strange. As depicted by you also, you only have it set for
183 and 200. You need a line like this:
onreply_route[1] {
if ((isflagset(5) || isbflagset(6)) &&
status=~"(100|180|183)|(2[0-9][0-9])") {
.........
}
Cheers,
DanB
On 9/4/07, Marc LEURENT <lftsy(a)free.fr> wrote:
I already have:
onreply_route[1] {
xlog("!!!! STARTING REPLY ROUTE\r\n");
#uac_restore_from();
if ((isflagset(5) || isbflagset(6)) &&
status=~"(183)|(2[0-9][0-9])") {
force_rtp_proxy();
}
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) {
fix_nated_contact();
}
exit;
}
???? Strange ?? no?
Dan-Cristian Bogos a écrit :
>> Hi Marc,
>>
>> this one should be handled in the same way u do with 183/200, catch it
>> in onreply route and force_rtpproxy()
>>
>> Cheers,
>> DanB
>>
>> On 9/4/07, Marc LEURENT <lftsy(a)free.fr> wrote:
>> 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.
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
>
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6289
days inactive
6289
days old
4 comments
2 participants
participants (2)
-
Dan-Cristian Bogos -
Marc LEURENT