{"id":2002,"date":"2008-07-29T10:01:32","date_gmt":"2008-07-29T14:01:32","guid":{"rendered":"http:\/\/blog.voipsupply.com\/?p=2002"},"modified":"2016-04-05T14:24:14","modified_gmt":"2016-04-05T14:24:14","slug":"setting-up-sip-ports","status":"publish","type":"post","link":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/setting-up-sip-ports\/","title":{"rendered":"Ask Mr. Andrews: Setting Up SIP Ports"},"content":{"rendered":"<p><strong>Dear Mr Andrews:<\/strong><\/p>\n<p><strong>Can you explain how to set SIP ports on modern popular hardware phones such as the medium priced (or are they entry level these days) Sipura\/Linksys\/Cisco line?<\/strong><\/p>\n<p><strong>Why would you not use 5060? If you have several phones behind NAT on the same LAN, is there a logical way to set these? How does the other endpoint see this? Enquiring minds and all that\u2026 I shall wait here on ICE for a STUNning discussion in a future article.<\/strong><\/p>\n<p><img loading=\"lazy\" decoding=\"async\" class=\"alignleft\" style=\"float: left\" src=\"\/blog\/voip-insider\/files\/2008\/07\/askandrews-logo2.gif\" alt=\"\" width=\"160\" height=\"109\" \/>Part of the problem with NAT is that there are several competing mechanisms for negotiating it, and the industry cannot seem to get behind a single, unified methodology.  Until this happens, dealing with NAT will likely continue to be a pain in the butt.<\/p>\n<p>STUN, ICE and TURN are three examples of solutions to issues inherent with SIP + NAT.  STUN is not 100% reliable depending on the type of NAT you are dealing with.  ICE builds upon STUN by allowing the device to use a range of ports and STUN techniques.  However, ICE is not well supported.  Media relay solutions like TURN can cause latency\/QoS issues with VoIP, and are generally difficult to scale.<\/p>\n<p>If you are behind NAT, you can set up port forwarding on your router\/firewall to allow VoIP traffic to pass through.  For SIP, use ports 5060 to 5070.  For RTP audio, use port 8766 to 35000.<\/p>\n<p>There are definitely some security concerns with port forwarding.  Hacking tools such as SIPFlanker http:\/\/tinyurl.com\/sipflanker are available as well as public posts detailing the default login credentials for many Sip devices http:\/\/tinyurl.com\/sippasswords<\/p>\n<p>When configuring Linksys devices behind NAT, there are a few things you want to be particular about.  In the configuration UI, in the \u201cSIP\u201d Tab, make sure you have the following options set:<br \/>\n*******************************************************************<br \/>\nSubstitute VIA Addr: yes<br \/>\nSTUN Enable: yes<br \/>\nSTUN Server:<\/p>\n<p>In the \u201cExt 1\u201d Tab, make sure you have the following options set:<\/p>\n<p>NAT Mapping Enable: yes<br \/>\n*******************************************************************<br \/>\nDouble check the NAT Keep Alive Interval setting on your Linksys phone and make sure it is set to a low value, ideally around 10-15 seconds.  For more information on configuring Linksys phones for NAT, refer to (starting) page 59 of the phone administration guide here http:\/\/tinyurl.com\/linksysspa.<\/p>\n","protected":false},"excerpt":{"rendered":"<p>Dear Mr Andrews: Can you explain how to set SIP ports on modern popular hardware phones such as the medium priced (or are they entry level these days) Sipura\/Linksys\/Cisco line? Why would you not use 5060? If you have several phones behind NAT on the same LAN, is there a logical way to set these? [&hellip;]<\/p>\n","protected":false},"author":7,"featured_media":0,"comment_status":"open","ping_status":"closed","sticky":false,"template":"","format":"standard","meta":{"footnotes":""},"categories":[1219],"tags":[],"class_list":["post-2002","post","type-post","status-publish","format-standard","hentry","category-voip-education"],"_links":{"self":[{"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/posts\/2002","targetHints":{"allow":["GET"]}}],"collection":[{"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/posts"}],"about":[{"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/types\/post"}],"author":[{"embeddable":true,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/users\/7"}],"replies":[{"embeddable":true,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/comments?post=2002"}],"version-history":[{"count":1,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/posts\/2002\/revisions"}],"predecessor-version":[{"id":164153,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/posts\/2002\/revisions\/164153"}],"wp:attachment":[{"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/media?parent=2002"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/categories?post=2002"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.voipsupply.com\/blog\/voip-insider\/wp-json\/wp\/v2\/tags?post=2002"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}