Can I avoid TURN ( Traversal Using Relays around NAT )?

[sarahr202]

Hi everybody

One of our customers uses hosted voip service . I just show two locations. Please see the attached picture that shows my truly artistic skills:)

Both locations have few phones and a router, all VOIP routing intelligence, call manger etc reside inside hosted VOIP service provider.

At each location, router performs NAT before forwarding and packet to VOIP provider. VOIP provider uses TURN to allow communication between the phones at both locations which are sitting behind NAT router.

So when the Phone in NY calls phone in PA, after all signalling, RTP stream flow via TURN server.between the phones.

I am thinking to avoid TURN server altogether . This is my thought process:

Let say  we remove the NAT at each location.

Suppose phone1 in NY calls phone2 in PA. After all signalling is done, Phone1 now knows the IP address of phone2 and vice versa thanks to SDP field in SIP message. Phone1 can now just send RTP stream to Phone  via routers i.e RTP does not need to go through relay  in VOIP provider.

is it possible to avoid TURN using my scenario ?

Much appreciated!!

[Vivek Batra · Accepted solution]

Hi,

If your question is "Can you avoid TURN in this case?", I would say "Yes, it's possible" if call control server is implemented accordingly.

However I don't see in either case to have RTP flows directly between the two phones since two phones are located behind the NAT router and you need media handling server to traverse RTP between two phones.

Coming back to the original question that can we avoid TURN? As per my experience and with most of the service providers what I saw.

1. With the help of SIP REGISTER, router makes the NAT UDP binding which allows service provider to send SIP INVITE to the phones located behind the NAT router.

2. With the help of symmetric signaling (rport) which allows service providers to send SIP responses (like 100 Trying, 180 Ringing etc) on the received IP address (layer3) instead of via header. In this way, the avoid NAT issues for SIP responses.

3. Service providers relay the RTP and also follow symmetric RTP which allows service provider to send RTP not on the IP address/Port which was communicated in SDP by phones, instead they send the RTP to the received IP address and port (means to NATed IP/port). In this way, even without using TURN and STUN, RTP bypass the NAT router and flows between two phones through service provider network.

Thanks

Vivek

[sarahr202]

Thanks