Hello,
I am sending in attach a pcap with a call between two webrtc clients, that
reproduces this scenario.
I am using as example the configuration at
https://github.com/havfo/WEBRTC-to-SIP .
I apply this change on the kamailio configuration to reproduce this
scenario:
@@ -350,6 +350,8 @@ request_route {
# authentication
route(AUTH);
+ sdp_keep_codecs_by_name("VP8","video");
+ msg_apply_changes();
# record routing for dialog forming requests (in case they are
routed)
# - remove preloaded route headers
remove_hf("Route");
Without this change, I can make calls between two webrtc clients.
Thanks for your help, so far.
Best regards,
Jose Lopes
On Mon, May 11, 2020 at 10:21 AM José Lopes <jose.lopes(a)itcenter.com.pt>
wrote:
Hello,
Sorry, I forgot to mention that, between the call of the two webrtc
clients, there is a B2BUA that only supports SIP UDP and RTP, so I need to
use rtpengine to translate from DTLS/SRTP to RTP.
I will try to make a call between the two webrtc clients and only use
kamailio without rtpengine to limit the issue.
Os melhores cumprimentos / Best regards,
*José Lopes*
Research and Development Technician
Phone: +351 256 370 980
Email: jose.lopes(a)itcenter.com.pt
On Tue, May 5, 2020 at 5:15 PM Juha Heinanen <jh(a)tutpro.com> wrote:
This 488 think reminds me that SIP over webrtc is
broken.
Webrtc UAS (at least JsSIP) cannot issue 488 before the UAS has started
to ring. It is very frustrating for the callee to get such a spam ring.
RFC3261 section "8.2 UAS Behavior" tells:
Note that request processing is atomic. If a request is accepted,
all state changes associated with it MUST be performed. If it is
rejected, all state changes MUST NOT be performed.
So it is against the standard to issue 180 and after that reject the
request with 488.
-- Juha
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