Hi,
After asking many questions, I haven't got any clues about how Kamailio handles INVITE message by default, in terms of modifying c= line in SDP
According to rtpproxy flow http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf
When client register, SIP proxy will call nat_uac_test() to detected if client is NATed or not, then save this info.
When client A calls client B, the INVITE message will go through SIP proxy. Here the SIP proxy can do 3 things (as in section "INVITEs behind NAT" in the pdf).
1. Add an SDP command direction:active to the SDP content 2. Change the c= line to a.b.c.d 3. Force RTP to go through a proxy by changing the c-line to c=IN IP4 address-of-proxy and the m-line to m=audio port-on-proxy RTP/AVP 0 101.
When will SER do 2, 3 ?
Hello,
On 3/6/13 1:54 PM, Khoa Pham wrote:
Hi,
After asking many questions, I haven't got any clues about how Kamailio handles INVITE message by default, in terms of modifying c= line in SDP
According to rtpproxy flow http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf
When client register, SIP proxy will call nat_uac_test() to detected if client is NATed or not, then save this info.
When client A calls client B, the INVITE message will go through SIP proxy. Here the SIP proxy can do 3 things (as in section "INVITEs behind NAT" in the pdf).
- Add an SDP command direction:active to the SDP content
- Change the c= line to a.b.c.d
- Force RTP to go through a proxy by changing the c-line to c=IN IP4
address-of-proxy and the m-line to m=audio port-on-proxy RTP/AVP 0 101.
When will SER do 2, 3 ?
2 and 3 are done usually when using rtpproxy to relay rtp packets (e.g., via rtpproxy_manage() function), or using various functions related to sdp updating from nathelper/rtpproxy/siputils or mangler module.
Cheers, Daniel
Hi Daniel, thanks for the response
2. Change the c= line to a.b.c.d .In the document, a.b.c.d is the public IP of the client !!!! so in this case not using rtpproxy 3. I read that Kamailio use rtpproxy when client is NATed. Does Kamailio detect NAT in REGISTER or INVITE message ?
I think this problem is very important but unfortunately, it lacks document :(
On Mon, Mar 11, 2013 at 7:52 PM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
Hello,
On 3/6/13 1:54 PM, Khoa Pham wrote:
Hi,
After asking many questions, I haven't got any clues about how Kamailio handles INVITE message by default, in terms of modifying c= line in SDP
According to rtpproxy flow http://kamailio.org/docs/ser-** getting-started/SER-**GettingStarted.pdfhttp://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf
When client register, SIP proxy will call nat_uac_test() to detected if client is NATed or not, then save this info.
When client A calls client B, the INVITE message will go through SIP proxy. Here the SIP proxy can do 3 things (as in section "INVITEs behind NAT" in the pdf).
- Add an SDP command direction:active to the SDP content
- Change the c= line to a.b.c.d
- Force RTP to go through a proxy by changing the c-line to c=IN IP4
address-of-proxy and the m-line to m=audio port-on-proxy RTP/AVP 0 101.
When will SER do 2, 3 ?
2 and 3 are done usually when using rtpproxy to relay rtp packets (e.g., via rtpproxy_manage() function), or using various functions related to sdp updating from nathelper/rtpproxy/siputils or mangler module.
Cheers, Daniel
-- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin
______________________________**_________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
hi i am so nubie about kamailio i follow the tutorial kamailio skype in one hour. i install on vps and give the public ip before i change the kamailio.cfg from the tutorial i run ps -fC kamailio have result and can register.but when i change kamailio.cfg from the tutorial run ps -fC kamailio no result and can't register. any help please need private sip server urgently
thanks