17 Sep
2014
17 Sep
'14
6:19 p.m.
Thanks Daniel, you save my day :-) Two rtpengine solves my problem and
works perfect. This solution also adds me the possibility to record calls,
when between two rtpengine instances I will put rtpproxy.
Regards,
Pawel
2014-09-17 9:35 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
Hello,
I would add the RTP-WebRTC gateway between SIP Kamailio and SIP UAC, from
resources point of view, it is the only leg that needs
encryption/decryption.
Otherwise, you can try to work with two rtpengine instances (sets) in WS
Kamailio, one to use for ws client to proxy and the other one for the leg
from proxy to ws client. It will be a communication between them with
classic rtp, both having towards ws client webrtc. It has the drawback of
decryption and encryption done two times for the same call. You would need
to add rtpengine set id in record-route to be able to handle properly the
re-INVITE, BYE, etc.
Another option that I would use is to send a negative reply from SIP
kamailio, catch that in failure_route in WS Kamailio and engace there the
rtpengine with proper flags. E.g., you assume it is going to be
webrtc-to-webrtc, so no encryption/decryption added first time invite comes
from WS client. You forward to SIP kamailio, which based on location, if it
discovers that the callee is classic SIP-RTP, will send a 4xx back to WS
Kamailio -- you end previous rtpengine session and engage it again with new
flags (use branch route for managing rtpengine -- like it is done in
default kamailio.cfg for rtpproxy).
Cheers,
Daniel
On 15/09/14 20:30, Paweł Sternal wrote:
Hi. Another topic about WebRTC, websockets with
kamailio and rtpengine ;-)
My problem is how to distinguish a call to WS UAC and how to SIP UAC in
scenarios:
1) WS client1 -> WS kamailio -> SIP kamailio -> SIP UAC
2) WS client1 -> WS kamailio -> SIP kamailio -> WS kamailio -> WS client2
WS kamailio is a proxy, SIP kamailio is a registrar
When "WS client1" is calling to "123123" WS kamailio doesn't know
if
"123123" was registered from "WS client2" or from SIP UAC.
I have in this case rtpengine_manage("....... RTP/AVP"), but when INVITE
is returned to WS kamailio? RTP/SAVPF?
Probably it is obvious, however...
When WS client2 reply with 200OK, rtpengine_manage("..... ICE=force") to
WS client1 SDP is sent without a:fingerprint. sipml5 dumps warning:
message: "Could not negotiate answer SDP; cause = NO_DTLS_FINGERPRINT
I tried different combinations... and I'm stuck :/
Regards
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Daniel-Constantin Mierla
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