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@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@itcenter.com.pt

On Tue, May 5, 2020 at 5:15 PM Juha Heinanen <jh@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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