Hello,
I am looking to get some feedback on some issues I noticed more and more
lately.
Apparently some SIP devices (media servers, phones, ...) are keeping the
"a=nortpproxy:yes" line in SDP when replying to an INVITE that contains
such line.
[Alice] ------> [Kamailio+RTPProxy] ------> [Bob]
The 200ok response from Bob has "a=nortpproxy:yes" in SDP.
By default, that line in SDP makes the rtpproxy not to engage itself
anymore in rtp relaying, and as a result things like no audio or one way
audio happens.
Anyone else encountering such situations? If yes, what are the devices
with such behaviour? So far I noticed with some FreeSwitch and Snom --
none of them I can control, so there might be a specific configuration
of those devices, not something by default there.
The solution is to set:
modparam("rtpproxy", "nortpproxy_str", "")
and use flag 'r' for rtpproxy_manage() if the IP in SDP is not a private
address.
I already updated the default config for master to use flag 'r' if the
SDP media IP is not private, wondering if nortpproxy_str should be set
to empty in kamailio.cfg (or made empty as default in config).
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat
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