[pjsip] PJSUA rtp nat traversal

Siram 56060 siram at blackngreen.com
Thu May 29 06:29:58 EDT 2014


Hi guys,

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??
Please help!!


Thank u,
Praveen
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140529/b21937d5/attachment-0002.html>


More information about the pjsip mailing list