Can you forward me the full invite coming in from the phone and I can
tell you why... what It feels like is they are doing crypto lines in
the RTP/AVP instead of RTP/SAVP as per the RFC says.
/b
On Apr 23, 2008, at 2:01 PM, Krzysiek wrote:
Hi
I have 2 softphones PhonerLite (they support SRTP via SDES ) and the
freeswitch (windows RC1 version) server and I wanted to make secure
call between those two endpoints (SRTP).
I spend whole day on testing this scenario and my conclusions are:
- when the option: <action application="export"
data="sip_secure_media=true"/> is uncommented, and both enpoints
have enabled SRTP then:
1) Initiator of the session sends SIP Invite with a=crypto paramter
and supported codecs
2) Freeswitch receives SIP Invite and sends SIP Invite to the
receiver (also with the crypto)
3) Receiver receives the SIP Invite with the a=crypto parameter and
he sends back supported codecs with 200 OK message (but without
a=crypto parametr. Is that ok? I'm afraid not)
4) Freeswitch sends 200 OK message but witout any codecs: m=audio 0
RTP/AVP 19 and no a= parameters!
5) Final result is that the second leg of the session between
Freeswitch and receiver has SRTP transport enbaled and the first leg
(initiator- Freeswitch) doesn't hear anything - no codecs! However
Freeswitch is sending RTP (not SRTP) pacekets to the initiator.
Could someone explain to me, what is going on, and why freeswitch
doesn't forward codecs accepted by the receiver to the initiator?
Is it a PhonerLite's bug or freeswitch's? Maybe someone has tested
SRTP with the PhonerLite softphone or any other free softphone with
srtp support?
When I uncommented: <param name="Inbound-no-media" value="true">
everything works fine. The parameter <action application="export"
data="sip_secure_media=true"/> doesn't change anything then (but i
cound miss something).
Thanks for help
Chris
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Brian West
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