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We are busy to develop an app to let users call eachother using webRTC. On this moment we have audio working, but video not. Now I hear that a proxy server can be the solution for this problem. But it seems a bit illogical, because what are the benefits to place a proxy server before the sip server Asterisk? Why should the video flow with a proxy but not without a proxy?

I heard of kamailio as a proxy, registrar server before asterisk. But I can't find any information about the benefits and why video stream will flow.

I hope someone can point me in some kind of direction to figure this out.

Thanks in advance.

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3 Answers 3

You haven't posted any addition details since asked for a couple of weeks ago so I'll try to answer as best I can (and recommend the question be closed):

Security: If your goal is to secure your Asterisk server then a proxy server / kamailio is not the answer. See Asterisk Security for details on how to do this.

Network: If your problem is NAT traversal then an SBC/Proxy can help - but isn't always necessary. Asterisk is pretty flexible in terms of publishing it's external IP in packets bound outside the local subnet. See NAT traversal for details on how to do this.

High Availability: If your goal is to add a proxy to redirect the traffic in the event of an Asterisk failure then a proxy isn't a great choice. (Nor is load balancing unless you don't need any session stickiness or synchronization etc). See High Availability for details on how to do this.

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A pure SIP proxy let you to manipulate all SIP message, this offers you a lot of fexibility and will help you dealing with NAT issues, failover and other feature not easy to implement with asterisk. Asterisk is a great and full-features PBX but doesn't offer many options for treating SIP messages ( this by design because is projected to be a PBX, not a SIP proxy).

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Kamailio will definitely not solve the problem with video, but it can offload some SIP processing from Asterisk, and add security. Also it allows to place multiple media servers behind one Kamailio server, thus allowing for load-balancing and failover.

I saw somewhere that recent Chrome update has broken compatibility with WebRTC implementation in FreeSWITCH, so probably asterisk is also affected. Also, video is only supported in Asterisk version 12.

I would actually suggest looking at FreeSWITCH and their mod_verto module. FreeSWITCH can replace Asterisk completely, and it's much easier to troubleshoot and integrate.

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Thanks for your input! I use asterisk 13, so I think videosupport is there, because video is one of the features of asterisk 13. Im getting errors of SDP; Error parsing/validating SDP body: Unknown error 220020. But I can't figure out how ineed to solve this, I think it is an chrome/client bug. –  Haije Ploeg Apr 1 at 8:56
    
You haven't posted enough information to help you solve the problem. Are you experiencing a protocol issue? Codec issue? Network/NAT issue? Can you post some data that will help us help you –  Telium Apr 2 at 0:03

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