Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

And are you forcing RTPengine to act as an ice light client? It looks like you are gettin a single ICE candidate in the answer back from freeswitch which would indicate that you are. I'd check your chrome webrtc statistics to see if tis failed to do do ice/stun negotiation on the 183. In

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

penSIPS users mailling list <users@lists.opensips.org> > *Subject:* Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc > > Actually the issue is i hear no audio on either side and just after > session progress (I guess when media starts coming from remote media > server

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

M > *To:* OpenSIPS users mailling list <users@lists.opensips.org> > *Subject:* Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc > > Actually the issue is i hear no audio on either side and just after > session progress (I guess when media starts coming from remote me

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

Take a look at the “fingerprint:” line. From: John Nash Sent: Thursday, June 23, 2016 3:42 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc Actually the issue is i hear no audio on either side and just after session progress (I guess when

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

the exchange, do this on offer and answer. From: John Nash Sent: Thursday, June 23, 2016 3:42 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc Actually the issue is i hear no audio on either side and just after session progress (I guess when

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

Actually the issue is i hear no audio on either side and just after session progress (I guess when media starts coming from remote media server) i see error "SRTP output wanted, but no crypto suite was negotiated" I had also checked media logs i could see RTP packets being sent from freeswitch

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the invite with an answer in the 183, and in the 200. What is the failure you are seeing, and where is it happening (in freeswitch? in the browser?)

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

OK here is the log https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 Sorry took me a while to convert wireshark trace to text file. My freeswitch is running on private IP (127.0.0.1) and opensips I run on both public and private so that for outside world opensips is the only

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

No - it's annoying to look at a trace that's had information removed and try and piece together whats happening. Your paranoid side is wrong, sorry. -Eric On 06/23/2016 01:06 PM, Patrick Wakano wrote: my paranoic side would recommend to hide/change private informations, specially any

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

my paranoic side would recommend to hide/change private informations, specially any authentication line that might appear... this is certainly a sort of social engineering threat we should worry... better be safe than sorry On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

I mean you can use a private gist, but you will be publishing the link in a public email list. In general I personally dont believe revealing ip addresses etc. is any problem - to put my money where my mouth is here is a gist link to an unaltered SIP trace on my server :)

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

Ok i am ready with logs. About gist may I use private option as traces have our IPs, user On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme wrote: > Hey John, > > Please paste a full UNALTERED sip trace into a gist (gist.github.com) > from the proxy servers perspective and provide

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

Hey John, Please paste a full UNALTERED sip trace into a gist (gist.github.com) from the proxy servers perspective and provide a link so that we can see what comes in, and what goes out from both sides. EG: ngrep -qtd any -W byline port 5060 This will show us the traffic that is leaving the

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

I double checked my rtpengine offer answer calls and now using https://github.com/onsip/sipjs-examples/tree/master/demo-phone but I face same issue (no audio either side) and error "SRTP output wanted, but no crypto suite was negotiated" Rtpengine also I updated to the latest now. Am I using

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is a much more active project that sipml5. 2. Im guessing that you are not properly passing flags to RTPEngine. If you want to have DTLS-SRTP between the browser, and plain RTP/AVP between RTPEngine and freeswitch, you need

[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

I am following http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying to test a call sipml5 --->Opensips + rtpengine > SIP end point (Freeswitch) But I do not have any audio on both sides. I see this error at rtpengine log "SRTP output wanted, but no crypto