Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Send my your cfg offlist. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/04/2017 07:07 PM, Alex Megalokonomos wrote: "You have to change a

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

"You have to change a bit the OpenSIPS script to move the offer and answer on 200 OK and ACK if the INVITE has no SDP attached." If you could provide some pointers on this that would be great. I'm guessing the t_on_reply ("handle_nat") stays as is While the branch_route[handle_nat] logic needs

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Yeah, sorry, missed that one . Well, it seems that OmniPCX is doing late SDP negotiation (via 200OK + ACK, instead of INVITE+200OK) and the tutorial script does not handle this case (for simplicity and clarity reasons). So, right now the RTPengine interaction (the offer and answer) are done

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

As you may have noticed in my last reply, I reached that far as well but got stuck later on on what appears to be the rtp engine configuration. Not strictly an Opensips issue but you might be able to help me. On Tue, Jul 4, 2017 at 6:07 PM, Bogdan-Andrei Iancu wrote: > Hi

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Hi Alex, Thank you for the offlist provided data. Shortly, the ACK received by OpenSIPS from OmniPCX is broken as it is missing all the Route headers. According to the pcap, it looks like: ACK sip:udoioiia@10.0.1.106:49246;transport=ws SIP/2.0 Record-Route:

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Hi Alex, As suspected, the ACK is not properly routed - see the retransmissions of the 200OK + ACK. SImply based on the logs I cannot see what the problem is - probably some missing fix_nated_contact() for the replies coming from the WS party. Please make a pcap capture + opensips log

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Good, there is some progress :). On the incoming calls, if the WS get's the call, we can park the part with the auth (it seems your opensips script is accepting calls from unknown sources...we can address this security hole later. So, if a call drop after 30 secs it usually means there is no

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

I think I set up uac_registrant correctly. I can dial out from a ws client and the ws extension rings from outside calls. However: a) on incoming calls, when ws client accepts, there is no sound and the line is dropped after 30 secs or so b) on outgoing calls, when the called extension accepts

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

I checked the script you mentioned and it does not help you - it has only UDP (no WS), it is really basic and it does not handle any REGISTER stuff, which is the trickiest - see https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/ or

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Hello Bogdan, First of all, thanks for your time. Unfortunately my SIP/OpensSIPS skills are what I've managed to learn in the last couple of days. I am a programmer but I've never had to work on SIP stuff before. Frankly to me, both solutions sound equally difficult since I have no idea where

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Hi Alex, To make a kind of WS<>UDP gateway you need a complete rework of the script presented in the tutorial, as it is a completely different SIP scenario. Not sure what are your SIP/OpenSIPS skills. But, there is a simpler alternative . Instead of a GW, you can make OpenSIPS as a

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Hello Bogdan, Yes, a gateway from WS to UDP (as well as DTLS-SRTP to RTP in order for it to work) is exactly what we're looking for. Unfortunately our Alcatel OmniPCX call center is a proprietary system that only allows for a limited number of SIP extensions (served from what appears to be an

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Hi Alex, First, some questions regarding the desired topology: 1) the WS end-points should register in OpenSIPS or all the way into Kamailio ? 2) also, the calls from the WS end-points should be all the time sent to Kamailio ? More or less, what I'm asking is : is OpenSIPS suppose to

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Curious, why would you want to use OpenSIPS and Kamailio? There both SIP proxies. Robert From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Alex Megalokonomos Sent: Wednesday, June 28, 2017 5:47 AM To: users@lists.opensips.org Subject: [OpenSIPS-Users] Opensips as SIP Proxy

[OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Hello, We have the following scenario: our office call center is an Alcatel OmniPCX Office setup. This handles most of our needs and also provides 4 SIP extensions. These are provided by what appears to be a Kamailio SIP server v 3.2.2 (no webrtc or websockets support) What we would like to do