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).
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 ?
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 -
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