Hello,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.
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).
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 ?
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.com
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