[OpenSIPS-Users] Strange behavior on ipv4 / ipv6 dual stack server with/without mhomed

[dlakelan at street-artists.orf]

On October 26, 2017 2:26:14 AM PDT, "Răzvan Crainea" <razvan at opensips.org> wrote:
>Hi, Daniel!
>
>1) I think you are absolutely right - if you have issues with phones 
>that misbehave when having IPv6 contact, topology_hiding is the 
>solution. There are several ways to determine whether an endpoint is 
>IPv6, so this should be easy to sort out.
>
>2) Have you considered any methods to inform Asterisk that your callee 
>is IPv6 or IPv4? Perhaps if you manage to figure out what the final 
>destination is, you can control the IP that Asterisk adds in SDP, and 
>prevent it from advertising IPv6.
>Also, did you consider using a media proxy? Something like RTPProxy set


Re figuring out in asterisk where to route call. I do not think this solution is easily possible due to mobile clients, those that can be sometimes v4 and sometimes v6. All my endpoints register to the proxy and not asterisk, so asterisk can not know ahead of time how they are connected. 

I suppose I could branch a register over to asterisk, but I think it's easier and more reliable to have opensips topo hide and rewrite the sdp. Asterisk with icesupport= yes will accept audio on any interface... It's a hack but a good option to keep moving part count down 😀

When it comes to adding media proxy, it seems this could work but also adds a component which Asterisk is capable of doing on it's own. If needed I will do this. Actually my favorite solution is to use  freeswitch instead of asterisk, and to run it on site at locations with v4 only atas etc. They then talk to the switch on site using private ips and the switch speaks to the internet and opensips across the NAT using dedicated port forwarding range for RTP or ipv6 as available. I'm considering a small SBC like raspberry pi to run freeswitch. Mobile clients with dual stack then simply register directly to my opensips on a VPS rather than through a local site freeswitch.

Another advantage is to offload transcoding from the VPS where asterisk currently runs. Timing issues in the VPS can be noticeable, particularly transcoding opus to g711 for the v4 atas to talk to mobile smartphones. Or mobile smartphones to POTS trunks. Another reason to prefer freeswitch, as it has fuller opus support.


-- 
Sent from my Android device with K-9 Mail. Please excuse my brevity.
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