[pjsip] SRTP feature requests

Klaus Darilion klaus.mailinglists at pernau.at
Mon Feb 25 11:17:54 EST 2008



Nanang Izzuddin schrieb:
> Hi Klaus,
> 
> Ticket 467 is already fixed and in the trunk.
> If you have a chance, please do some tests and report back the result.

Hi Nanang!

I just made some test and now pjsua automatically hangs up  the call, but:

1. The BYE is sent before the ACK (from INV-200). I think this is valid 
but IMO it would be nicer if BYE is sent after ACK.

2. The BYE contains a session description, this is IMO a bug:

BYE sip:klaus.darilion at 10.10.33.21:5060 SIP/2.0
Via: SIP/2.0/UDP 
10.10.0.51:3891;rport;branch=z9hG4bKPj096a048772764b659103cb06bc9b80ff
Max-Forwards: 70
From: sip:klaus.darilion at nic.at43.at;tag=2e731928dd334b91b716f476058249ac
To: sip:klaus.darilion at nic.at43.at;tag=000cce3a7bf800037b595478-6250939a
Call-ID: e20d9cd2582347d19032264cf02b5461
CSeq: 5060 BYE
Route: <sip:10.18.53.113:6060;lr>
Route: <sip:10.10.32.160;lr;ftag=2e731928dd334b91b716f476058249ac>
Content-Type: application/sdp
Content-Length:   418

v=0
o=- 3412948447 3412948447 IN IP4 10.10.0.51
s=pjmedia
c=IN IP4 10.10.0.51
t=0 0
m=audio 4002 RTP/AVP 103 102 104 117 3 0 8 101
a=rtcp:4003 IN IP4 10.10.0.51
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


regards
klaus




> Thanks.
> 
> cheers,
> nanang
> 
> 
> On 14/02/2008, *Nanang Izzuddin* <nanang.izzuddin at gmail.com 
> <mailto:nanang.izzuddin at gmail.com>> wrote:
> 
>     Hi Klaus,
> 
>     No.1 is added to the existing ticket, I think the cases are related:
>     http://trac.pjsip.org/repos/ticket/467
> 
>     No.2 should be good and useful.
>     https://trac.pjsip.org/repos/ticket/479
> 
>     Thanks for the reports (& request :D)
> 
> 
>     nanang
> 
> 
> 
>     On 14/02/2008, Klaus Darilion <klaus.mailinglists at pernau.at
>     <mailto:klaus.mailinglists at pernau.at>> wrote:
>      > Hi!
>      >
>      >  I did some SRTP tests and it mostly works, but:
>      >
>      >  1. It does not work if pjsip sends SAVP and 200 OK contains AVP
>     - then
>      >  pjsip has an open call but does not send RTP. The logs say:
>      >
>      >   16:02:54.437   pjsua_call.c SDP negotiation has failed: SDP media
>      >  transport type mismatch in offer/answer (PJMEDIA_SDPNEG_EINVANSTP)
>      >  [status=220046]
>      >
>      >  What should happen now? Should pjsua-lib hang up the call? It
>     does not
>      >  and the call changes to "confirmed".
>      >
>      >  If the application should send the BYE, then there should be some
>      >  indication, e.g.     PJSUA_CALL_MEDIA_SAVP_FAILED in the
>      >  call_media_state callback. Or is there already a callback which
>     tell the
>      >  application about the SRTP failure?
>      >
>      >  2. If SRTP is optional and a call is established, it would be
>     good to
>      >  retrieve SRTP status from pjsua (e.g. pjsua_call_get_info()) to
>     indicate
>      >  SRTP status to the user.
>      >
>      >  thanks
>      >  Klaus
>      >
>      >
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> 
> 
> 
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