[Spce-user] sipwise 3.1 (CE) and websocket support

[Ashutosh Apte]

Hi Andrew,
    Thanks for your reply. I made the recommended changes and now I can see
that the INVITE request received by the legacy gateway does contain a
RTP/AVP media profile. I've run into the next problem though.

The legacy gateway does not receive an ACK so the call never gets
established. (I do see that media from the WebRTC client does make it all
the way to the legacy gateway so that's a good news.)

After about 32 seconds (while the gateway is re-transmitting its 200 OK
response), ngcp sends an ACK (it is internally generated) followed by a BYE
request to end the call.

I see the following errors in the kamailio-proxy.log:

-----
Feb 14 20:41:58 spce proxy[3785]: DEBUG: <script>: start of route
ROUTE_OUTBOUND - 1014 ACK
Feb 14 20:41:58 spce proxy[3785]: WARNING: <core> [msg_translator.c:2499]:
via_builder(): *TCP/TLS connection (id: 0) for WebSocket could not be found*
Feb 14 20:41:58 spce proxy[3785]: ERROR: <core> [msg_translator.c:1725]:
build_req_buf_from_sip_req(): could not create Via header
Feb 14 20:41:58 spce proxy[3785]: ERROR: <core> [forward.c:607]:
forward_request(): ERROR: forward_request: building failed
Feb 14 20:41:58 spce proxy[3785]: ERROR: sl [sl_funcs.c:371]:
sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server
error occurred (1/SL)
Feb 14 20:41:58 spce proxy[3785]: DEBUG: <script>: exit of route
ROUTE_OUTBOUND - 1014 ACK
-----

Note that I'm using *ws://*172.18.101.48:5060/ws (and not http://) as the
websocket server URL. Andreas had suggested earlier to use *http:*//ip:port/ws.
The WebRTC clients that I'm using for testing (sipML5 and jsSIP) do not
like anything other than ws:// for Websocket server URL.

Not sure if this is the real issue as the request fails to get sent to the
legacy gateway, which is not using Websockets but the above mentioned error
occurs when it is ready to send the ACK to the legacy gateway.

Also attached is a .zip file containing the proxy and lb logs.

Thanks again for all your help. I think I'm getting closer to get
everything working.

Ashutosh



On Fri, Feb 14, 2014 at 1:38 AM, Andrew Pogrebennyk <
apogrebennyk at sipwise.com> wrote:

> On 14/02/14 01:33, Ashutosh Apte wrote:
> > I've also created a SIP Peering Group to route calls to a legacy SIP
> > gateway. On dialing, the call hits the legacy gateway but I notice that
> > the SDP body in the INVITE message uses RTP/SAVF. This gateway does not
> > support sRTP. Is there a way to force the mediaproxy to use RTP only?
> >
> > I've changed the parameter under Domains -> Preferences -> NAT and Media
> > Flow Control -> srtp_transcoding from "transparent" to "ForceRTP". The
> > kamailio proxy debug logs do show that the parameter was indeed changed
> > but it did not have any effect on the INVITE message.
> >
> > What configuration parameter should I be changing to get this to work?
>
> Hi,
> from my PoV you should set srtp_transcoding in the domain to "Prefer
> SRTP" (AFAIK it doesn't work if you leave it as "transparent" and I want
> to add automatic detection of endpoints that use RTP/SAVP and RTP/SAVPF)
> and most important - srtp_transcoding in the peer preferences (for your
> legacy gateway) to "Force RTP". This will enable transcoding. You may
> also need to set rtcp_feedback to "Prefer AVPF" on the domain and "Force
> AVP" on the legacy gateway.
>
> Andrew
>
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