Ok, I may have a look to the csv.
Thanks for the help.
regards,
Nicolas
Bogdan-Andrei Iancu wrote:
Hi,
Nicolas Olivier wrote:
>
> Hi Bogdan,
>
> Ok, I understand now. But I still encounter the problem because:
> - rtpproxy only rewrites the c= from media part (but it should be fine
> as you said) despite what a quick look in the rtpproxy code comments
> say ("We have to change ports in m-lines, and also change IP addresses
> in c-lines which can be placed either in session header (fallback for
> all medias) or media description.")
yes, the nathelper will change the c= from session header only if it
finds a media section without a local c= (which means the default c=
from session hdr will be used).
> - the centrex (which is an asterisk by the way) take only into account
> the c= from the session part, not the one from media part
in the CVS devel there is a flag that force also changing of session c= :
http://openser.org/docs/modules/1.1.x/nathelper.html#AEN275 , the
"c" flag
regards,
bogdan
>
>
>
> regards,
> Nicolas
>
> Bogdan-Andrei Iancu wrote:
>
>> Hi Nicolas,
>>
>> it;s perfectly ok - see the SDP RFC : an SDP may contain a default
c= in
>> the session part; each media section (m=) may contain an ip (c=); if it
>> doesn't the session c= will be used.
>>
>> regards,
>> bogdan
>>
>> Nicolas Olivier wrote:
>>
>> >
>> > Hi,
>> >
>> > I've got a gateway which is only used for rounting and rtp proxying
>> > between providers and centrexes.
>> >
>> > On reply to an INVITE, one of our provider send back a "183 Session
>> > Progress". The problem is that in the SDP block, we've got 2
media IP
>> > address and rtpproxy only rewrite one.
>> >
>> > Finally, the provider establish rtp session with our gateway, and
our
>> > centrex directly with the provider.
>> >
>> > provider gateway centrex
>> > 172.16.0.10 192.168.1.10 192.168.1.20
>> > RTP -------------> RTP ------------> RTP
>> > ^-------------------------------------------------|
>> >
>> > So my questions are, is it possible to have multiple IP address in
>> SDP
>> > and if so, how can I tell rtpproxy to rewrite all of them.
>> >
>> > Coming from provider:
>> >
>> > SIP/2.0 183 Session Progress.
>> > Via: SIP/2.0/UDP
>> > 192.168.1.10;branch=z9hG4bKdd67.a4cc2c44.0,SIP/2.0/UDP
>> > 192.168.1.20:5062;branch=z9hG4bKdd67.08f45a33.0,SIP/2.0/UDP
>> > 192.168.1.20:5060;branch=z9hG4bK4af242b7.
>> > From: "02" <sip:[EMAIL PROTECTED]>;tag=as226ce7b9.
>> > To: <sip:[EMAIL PROTECTED]:5062>;tag=3123AAA8-20C5.
>> > Date: Tue, 11 Apr 2006 09:10:29 GMT.
>> > Call-ID: [EMAIL PROTECTED]
>> > Server: Cisco-SIPGateway/IOS-12.x.
>> > CSeq: 102 INVITE.
>> > Allow-Events: telephone-event.
>> > Contact: <sip:[EMAIL PROTECTED]:5060>.
>> > Record-Route:
>> >
>>
<sip:192.168.1.10;ftag=as226ce7b9;lr=on>,<sip:192.168.1.20:5062;ftag=as226ce7b9;lr=on>.
>>
>> >
>> > Content-Disposition: session;handling=required.
>> > Content-Type: application/sdp.
>> > Content-Length: 261.
>> > .
>> > v=0.
>> > o=CiscoSystemsSIP-GW-UserAgent 3448 4768 IN IP4 172.16.0.10.
>> > s=SIP Call.
>> > c=IN IP4 172.16.0.10.
>> > t=0 0.
>> > m=audio 18322 RTP/AVP 18 101.
>> > c=IN IP4 172.16.0.10.
>> > a=rtpmap:18 G729/8000.
>> > a=fmtp:18 annexb=no.
>> > a=rtpmap:101 telephone-event/8000.
>> > a=fmtp:101 0-16.
>> >
>> > Forwarded to centrex:
>> >
>> > SIP/2.0 183 Session Progress.
>> > Via: SIP/2.0/UDP
>> > 192.168.1.20:5062;branch=z9hG4bK43a4.3e96aba3.0,SIP/2.0/UDP
>> > 192.168.1.20:5060;branch=z9hG4bK3213db83.
>> > From: "02" <sip:[EMAIL PROTECTED]>;tag=as1a2f900d.
>> > To: <sip:[EMAIL PROTECTED]:5062>;tag=3121D1B4-1BFD.
>> > Date: Tue, 11 Apr 2006 09:08:28 GMT.
>> > Call-ID: [EMAIL PROTECTED]
>> > Server: Cisco-SIPGateway/IOS-12.x.
>> > CSeq: 102 INVITE.
>> > Allow-Events: telephone-event.
>> > Contact: <sip:[EMAIL PROTECTED]:5060>.
>> > Record-Route:
>> >
>>
<sip:192.168.1.10;ftag=as1a2f900d;lr=on>,<sip:192.168.1.20:5062;ftag=as1a2f900d;lr=on>.
>>
>> >
>> > Content-Disposition: session;handling=required.
>> > Content-Type: application/sdp.
>> > Content-Length: 277.
>> > .
>> > v=0.
>> > o=CiscoSystemsSIP-GW-UserAgent 565 174 IN IP4 172.16.0.10.
>> > s=SIP Call.
>> > c=IN IP4 172.16.0.10.
>> > t=0 0.
>> > m=audio 36296 RTP/AVP 18 101.
>> > c=IN IP4 192.168.1.10.
>> > a=rtpmap:18 G729/8000.
>> > a=fmtp:18 annexb=no.
>> > a=rtpmap:101 telephone-event/8000.
>> > a=fmtp:101 0-16.
>> > a=nortpproxy:yes.
>> >
>> >
>> > openser.cfg
>> >
>> > (...)
>> >
>> > onreply_route[1] {
>> > if (status =~ "(180)|(183)|2[0-9][0-9]") {
>> > fix_nated_contact();
>> > if (!search("^Content-Length:[ ]*0")) {
>> > force_rtp_proxy();
>> > };
>> > } else if (nat_uac_test("1")) {
>> > fix_nated_contact();
>> > };
>> > }
>> >
>> > (...)
>> >
>> > Best regards,
>> > Nicolas Olivier
>> >
>> >
>> > _______________________________________________
>> > Users mailing list
>> > [email protected]
>> > http://openser.org/cgi-bin/mailman/listinfo/users
>> >
>>
>
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