[pjsip] PJSUA rtp nat traversal
siram at blackngreen.com
Thu May 29 06:29:58 EDT 2014
I've been trying to make a sip server using PJSUA, and using linphone and
vimphone clients to test it.
After some efforts, by modifying the messages on_rx_data and on_tx_data i
could pass the sip signalling messages properly
But after the states going to "confirmed" state, i could hear audio
(bidirectional) only if client (using stunserver) makes a call to server.
When server tries to make a call to the registrar fetched address of the
client, i find only upbandwidth in client but no downbandwidth. Auto RTP
switching feature ( PJMEDIA_UDP_NO_SRC_ADDR_CHECKING) is turned on (set to
0) in PJMEDIA_TRANSPORT_UDP_ATTACH api
How to overcome this??
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