[SIP SIMPLE client] beginner SylkServer hiccups
jpyle at fidelityvoice.com
Tue Apr 26 15:06:55 CEST 2011
Sipclient list. Got it.
How does one "just add a cert"? What should the filenames be called? What format should the files be in? Should one follow the Opensips-style procedure for creating them, the Mediaproxy procedure (tinyca), or something else? What procedure can one follow to create certs for the clients, or can the client use use the same cert as the server?
I don't necessarily need to use TLS in my environment. Especially on this local LAN for testing. TCP would be just fine. How do I get SylkServer to populate the SDP with the correct interface's IP in the MSRP line of the SDP when not using TLS? At the moment it's using the public IP of the box (the one with the default route) instead of the private one on the interface where its SIP stack is listening.
From: Adrian Georgescu <ag at ag-projects.com<mailto:ag at ag-projects.com>>
Date: Tue, 26 Apr 2011 02:23:31 -0400
To: Jeff Pyle <jpyle at fidelityvoice.com<mailto:jpyle at fidelityvoice.com>>
Cc: "sipclient at lists.ag-projects.com<mailto:sipclient at lists.ag-projects.com>" <sipclient at lists.ag-projects.com<mailto:sipclient at lists.ag-projects.com>>, "blink at lists.ag-projects.com<mailto:blink at lists.ag-projects.com>" <blink at lists.ag-projects.com<mailto:blink at lists.ag-projects.com>>
Subject: Re: [SIP SIMPLE client] beginner SylkServer hiccups
On Apr 26, 2011, at 2:47 AM, Jeff Pyle wrote:
The Contribute page on the SylkServer website refers to the Blink mailling list, while the Contact page references the SIP Client one. Which one is preferred?
Sipclient. Page is fixed.
Referring to the first posting quoted below… The presence or absence of audio didn't have anything to do with the transport of the SIP packets, but rather the tls or tcp setting of the MSRP Transport in Blink. A setting of tcp causes an msrp:// URI in Blink's INVITE, and a corresponding msrp:// URI in SylkServer's 200 OK. A setting of tls causes msrps:// URIs in both cases. This makes sense. What does not make sense is in the tcp case, the URI from SylkServer contained the box's public IP (the wrong interface), while in the tls case the URI contained the (private) IP of the Blink client. In effect it mirrored the URI of the INVITE. Or, does this make sense and I'm simply not aware of the underlying logic?
Just add a cert and use TLS.
In both cases the c= line was correct for the private IP of the corresponding UA, yet Blink only recognized the audio when the MSRP transport was set to tls. I verified the IP addresses and ports of the RTP and RTCP packets were as described in the INVITE and 200 OK. This one doesn't make sense to me either.
Even when the tcp setting was used in Blink, and the URI contained the public IP of the SylkServer on port 2855, I still didn't see the SylkServer machine listening on any interface on port 2855.
A nudge towards some documentation would be most helpful.
From: Jeff Pyle <jpyle at fidelityvoice.com<mailto:jpyle at fidelityvoice.com>>
Date: Mon, 25 Apr 2011 17:42:24 -0400
To: "sipclient at lists.ag-projects.com<mailto:sipclient at lists.ag-projects.com>" <sipclient at lists.ag-projects.com<mailto:sipclient at lists.ag-projects.com>>
Subject: [SIP SIMPLE client] beginner SylkServer hiccups
I am attempting to deploy SylkServer behind an Opensips instance on a local LAN for testing purposes. My progress is mixed.
Sylk's config.ini is completely default with the exception of moving the tcp and udp ports from 5060 to 5080. conference.ini is completely default.
With 100% udp transport, I'm able to connect an audio conference session from Blink through Opensips to SylkServer. When I try to use the chat window, I receive back a 488 Not Acceptable Here.
My thought was perhaps I need tcp for this to work. IM over SIP, including MSRP, is new to me. I converted the Blink-Opensips and Opensips-Sylkserver communications to TCP. Now, the audio conference session appears to work, including RTP in both directions, I just can't hear it on Blink. And Blink disconnects the session with a BYE after a few seconds.
I'm flying a bit blind here. Is there documentation for SylkServer, other than what's contained in the default configurations? Do I need anything other than SylkServer to handle the MSRP session? It's my understanding Sylk will do that, but I can't get it to listen on port 2855.
Any direction towards some documentation would be greatly appreciated.
SIPclient mailing list
SIPclient at lists.ag-projects.com<mailto:SIPclient at lists.ag-projects.com>
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the SIPBeyondVoIP