Running SIP behind NAT Dr. Christian Stredicke, snom technology AG, Miami, USA, February 2002

V1.0 2 Some Statistics First… Who made a cell phone call here? Who made a SIP call here? –Who tried? –Who succeeded? Who has a SIP phone at home? The Pulver Report - January 16, 2003 Issue: –“2003: The Year of Consumer Communications“ –Looks like we have to change a couple of things

V1.0 3 Which information does a client has to set up for port forwarding in NAT equipment? Router needs information where to send packets in private network –Map port to private address and port –By default packets will be rejected or sent to DMZ Router needs hint for security checking –Accept packets from any destination –Accept packets only from associated host –Accept packets only from associated host and port Router Client

V1.0 4 How did other applications solve the problem? HTTP, telnet, … –Using TCP DNS, others –“Digging holes”: Set up association when client sends out packet from unmapped port for seconds –Security policy hardwired by vendor –Some offer a DNS proxy (application layer gateway) ftp –Does not work! –Inexperienced users use http instead –Some routers offer applications layer gateway Heterogeneous environment –Every vendor does it in a different way –“Digging holes” is common denominator

V1.0 5 STUN uses the digging hole trick to set up port associations Initialization procedure checks environment –Goal: Check if STUN is needed –Type of NAT does actually not really matter because user is not interested in failure reason SIP port kept alive by sending packets every s RTP ports are allocated dynamically when starting a call –Otherwise keep-alive traffic would be double –RTCP port can not be allocated because next port allocation is unlikely –Long ringing and putting caller on hold is problematic (no port refresh during this time)

V1.0 6 TURN works in symmetrical NAT environment, but has too many problems Scalability –TURN server becomes “media server” –Every call generates about 50 packets per second Delay –Sending packets over media server increases transport delay significantly –E.g. local call in Tokyo when TURN server is in Frankfurt TURN specification –Needs rework (activation message not defined)

V1.0 7 UPnP is the right approach. Generic protocol to allocate ports on router –Works with SIP, can be used with other applications as well –Can be integrated with firewalls –Not too hard to implement Microsoft Messenger uses UPnP –“De facto standard” –Many DSL router vendors offer UPnP now Problem: Old Equipment –Use STUN –Maybe use TURN, even if call duration is terrible –Instruct customers to set up ports manually

V1.0 8 The 90 % Problem: STUN works fine in 90 % of the cases Some routers do not run STUN without user interaction –Stateful inspection –Trying to be smart –Users must set up DMZ 10 % support calls are intolerable STUN can only be „gap-filler“ –“Best Effort” –No support Need clear indication if VoIP will work –Clear technical specification under which circumstances customers can expect setup to work –UPnP is good candidate for this

V1.0 9 With the increasing availability of UPnP, most home customers can be addressed UPnP STUN UPnP STUN End of 2002End of 2003 Software Updates New Equipment

V Application layer gateways (ALG) solve the problem in the business area Business customers have different requirements than home users –Many phones –Want to run proxies, media servers, application servers behind their firewall –These applications probably will not have UPnP or STUN Therefore, firewalls will probably include SIP-aware ALG Sample SIP NAT ALG available from snom –No support! Commercial products e.g. from Cisco, Intertex, Ingate, Jasomi, …

V Calling phones in the same network requires ancillary information* 1a) Phone A sends to public address of B 1b) Router will not forward packet, call will fail 2) A knows B is in the same NAT and sends packet to private address instead * If no ALG is involved

V Ancillary information must be placed in contact URI and in SDP* INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK-6rms4e9tmtsz Max-Forwards: 70 From: ;tag=16z5zw9lqt To: Call-ID: CSeq: 1 INVITE Contact: Content-Type: application/sdp Content-Length: 311 v=0 o=root IN IP s=SIP Call c=IN IP t=0 0 m=audio RTP/AVP a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp: a=x-private: : :10004 * Non-standardized example

V NAT2NAT3 NAT1 Multi-tier NAT requires a list of private addresses and a STUN/UPnP server between the NATs A has three identities: : : :5678 B has three identities: : : :5679 STUN Phone APhone B When using STUN, a STUN server is required between the layers

V How should a phone boot up? Try UPnP Use UPnP Try to Register Use STUNUse Given Identity UPnP available No response (5 seconds) or not available No problem: either public address, ALG or total private environment Registrar complains about private address This step can be done even without STUN, as the registrar returns the response quick

V Is UPnP secure? A possible man-in-the-middle attack scenario… 1. A opens RTP forwarding port Phone BPhone A 2. B retrieves forwarding table 3. B rearranges port forwarding 4. B receives all RTP from the IAD and forwards it to A (after recording it) Same attack can be done with signaling Can be solved with TLS and SRTP

V Security is ok for home networks, but for business networks some enhancements are needed How much security needs a home? –Son listens to call of daughter –Son listens to call of father doing telephone banking –Son using packet sniffer, son is listening on the door STUN is also not secure –ARP attacks can also redirect the packet flow (however that’s not so easy) Attacks from the outside –Orphan bindings may give access to private devices –Devices should be able to deal with this anyway Security enhancements in UPnP Version 2 Businesses should use ALG which takes care about it

V To make UPnP more reliable, clients need to allocate bandwidth Don’t allocate bandwidth “just in case” –Allocating ports in the beginning is easy and can set scheduling priorities –But when too many VoIP calls are done, all of them suffer Ask for bandwidth before a call starts –Sending busy is better than having stuttering calls –Phone needs to know when bandwidth is available again so that call completion can be indicated –Notification when bandwidth is available Could be added to current allocation requests –Bandwidth indication –Insufficient bandwidth as denial reason

V Conclusion: You must choose what to tell the customer If you can, use an ALG –Works will all SIP-compliant equipment –Most expensive solution, but complete functionality Else if you can, use UPnP –Works with all SIP- and UPnP-compliant equipment –“MS Messenger” solution, routers for 65 $ available –Problems making calls within the private network Else if you dare, use STUN –Works with all SIP- and STUN-compliant equipment if the routers are not inspecting packets –Could become support-headache –Also problems in the private network If you also want to support the rest, think about TURN –Works with all SIP-, STUN/TURN-compliant equipment and the 99% of the NAT routers

© 2003 snom technology Aktiengesellschaft Written by: Dr. Christian Stredicke Version: 1.0 The author has made his best effort to prepare this document. The content is based upon latest information whenever possible. The author makes no representation or warranties of any kind with regard to the completeness or accuracy of the contents herein and accept no liability of any kind including but not limited to performance, merchantability, fitness for any particular purpose, or any losses or damages of any kind caused or alleged to be caused directly or indirectly from this document. For more information, mail Pascalstr. 10E, Berlin, Germany.

V In cases when NAT is symmetrical, TURN could be a solution Router Client STUN/TURN Server Allocate Request/Response 2. Activate Request/Response 3. SIP/Media