29 Jun
2021
29 Jun
'21
7:43 p.m.
Thanks Muhammad,
That seems to have done the trick. I am now able to bridge WebRTC to UDP.
Regards,
Peter Manley
M S shaheryarkh at
gmail.com<mailto:sr-users%40lists.kamailio.org?Subject=Re%3A%20%5BSR-Users%5D%20Guidance%20building%20a%20WebRTC%20to%20Asterisk%20Intermediate%0A%20Proxy&In-Reply-To=%3CCAFZQphyZE_tGB9%3D5tFQBtXAv5twMh%3D9aCNFAvveGf0-L%3Dy_qwQ%40mail.gmail.com%3E Sat Jun 26 03:57:27 CEST 2021
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________________________________
The problem seems to that the kamailio is trying to route it over UDP port
using WSS transport, which will never work.
You are creating a transport bridge (UDP <-> WSS i.e. stateless transport
<-> stateful transport) as well media bridge (RTP <-> SRTP), so it is going
to be tricky. I strongly discourage it since things get messy for
sequential requests. But you can try the following,
Here are some PATH module parameters and functions, you will need to setup.
1. You will need to enable "use_received" parameter to ensure connection
tracking.
2. It seems you may have enabled "received_format", so you make sure you do
NAT detection and use "handle_ruri_alias" for initial requests.
3. You must also enabled "enable_r2", but with caution, since both ends
(i.e. UAC and UAS) can be using same kamailio (proxy) socket for in/out.
4. Make sure to use "add_path_received" instead of "add_path".
5. If using OUTBOUND module then make sure it is loaded BEFORE the PATH
module.
Thank you.
--
Muhammad Shahzad Shafi
Burraq Technologies
Tel: +49 176 99 83 10 85
On Sat, Jun 26, 2021 at 12:47 AM Peter Manley <peter.manley at
kombea.com<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users&…
wrote:
> SIP UAC < ---UDP/SIP --- > Kamailio/RTPEngine Proxy < ---UDP/SIP-----
> WebRTC UAC < ---WSS--- > Kamailio/RTPEngine Proxy < ---UDP/SIP-----
> Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core
> Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core
> Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core
> Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core
> www.kombea.com<http://www.kombea.com
>
<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-g&c=aXfg9j-An5sFdflqnjAj_USiCjXm5xEBIX8lTgWEt0U&r=ykVOlDeeWPn-JjTs8avieNrFskQ3kNzjf6_Jn_wG5HLeLIRsvzCN4-dJx6BFR4el&m=MMSIepRfj2CXVMH3CH8SghyPAoonRISMH__5bcNFPLM&s=ocueS1ZL34IN6qr14M-aItGjeJGQ2km5U2zpzO4Fyqw&e=
Peter H. Manley, Ph.D.
Chief Software Architect
KomBea Corporation
3400 North Ashton Blvd, Suite
470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com… Lehi, Utah
84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.c…
www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.k…
Office:+1.801.610.5808
Mobile +1.801.243.7773
Hello,
I'm working on building an RTP Proxy which will
allow both regular SIP to
SIP transactions and WebRTC to SIP transactions to an
Asterisk Server.
Asterisk Server
AND
Asterisk Server
The Kamailio/RTPEngine Proxy is stateless (does not
use the tm or
registerer modules), and uses the Path module to
replace the Route header
with a Path header.
It works grep for SIP/UDP to SIP/UDP transactions.
HOWEVER... It only partially works for WSS to SIP/UDP
transactions. The
REGISTER/401(Unauthorized), REGISTER/200(OK)
transaction works well.
When the Asterisk server sends a SIP Options message,
I get the following
error:
Jun 25 15:47:03 VRTPENGINE kamailio[4953]: INFO:
<script>: => OPTIONS <
> sip:l4umfumh at
192.0.2.97<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users&…
- <null> (IP:172.16.0.48:sp)
[core/msg_translator.c:2815]: via_builder(): TCP/TLS
connection (id: 0) for
WebSocket could not be found
[core/msg_translator.c:2003]:
build_req_buf_from_sip_req(): could not
create Via header
[core/forward.c:550]: forward_request(): building
failed
[core/receive.c:425]: receive_msg(): error while
trying script
It appears that the WebSocket corresponding to the WSS
leg cannot be
located when the request is initiated from the
Asterisk Server.
Any thoughts on how I can get this to work in the
other direction
(Asterisk to Kamailio)? I've been looking at the
WebRTC to SIP Client and
Server (https://github.com/havfo/WEBRTC-to-SIP), but
that doesn't quite
do what I want it to do (i.e. it appears to act as a
registrar server and
doesn't forward REGISTER requests on to another
registrar).
Thanks and regards,
*Peter H. Manley, Ph.D.*
Chief Software Architect
KomBea Corporation
3400 North Ashton Blvd, Suite 470
>
<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3D3400-2BNorth-2BAshton-2BBlvd-2C-2BSuite-2B470-2B-250D-250A-2BLehi-2C-2BUtah-2B84043-26entry-3Dgmail-26source-3Dg&d=DwMF-g&c=aXfg9j-An5sFdflqnjAj_USiCjXm5xEBIX8lTgWEt0U&r=ykVOlDeeWPn-JjTs8avieNrFskQ3kNzjf6_Jn_wG5HLeLIRsvzCN4-dJx6BFR4el&m=MMSIepRfj2CXVMH3CH8SghyPAoonRISMH__5bcNFPLM&s=C4toC0HQ2JwHKiamawbqvdjdidlwyhCP0XVneFagwkY&e= Lehi, Utah 84043
>
<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3D3400-2BNorth-2BAshton-2BBlvd-2C-2BSuite-2B470-2B-250D-250A-2BLehi-2C-2BUtah-2B84043-26entry-3Dgmail-26source-3Dg&d=DwMF-g&c=aXfg9j-An5sFdflqnjAj_USiCjXm5xEBIX8lTgWEt0U&r=ykVOlDeeWPn-JjTs8avieNrFskQ3kNzjf6_Jn_wG5HLeLIRsvzCN4-dJx6BFR4el&m=MMSIepRfj2CXVMH3CH8SghyPAoonRISMH__5bcNFPLM&s=C4toC0HQ2JwHKiamawbqvdjdidlwyhCP0XVneFagwkY&e= Office:+1.801.610.5808
Mobile +1.801.243.7773
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Peter Manley