we are testing ipv6 sip calling through sip proxy (asterisk) and we
are trying to redirect audio stream directly between IPv6 endpoints
(Linphone). Both endpoints are linux running dual stack but for the
call establishemente we are using the ipv6 network. the endpoints also
have ipv4 conectivity
Linphone seems to be incorrectly handling sip reinvites issued by the
proxy. Linphone always replies to a reinvite with a 200/ok where the
SDP message connection information field contains: IN IP4 127.0.0.1
instead of containing: IN IP6 <IPv6 address of linphone sending 200/ok>
Anyway the call is correctly established with the RTP strem going
through the proxy server, instead of going directly between the
endpoints. (In IPv4-IPv4 linphone calls, correct reinvite 200/ok
answers are sent, and the audio is redirected)
Any idea if this is a bug?
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