How does SIP phone behind firewall work in a hosted VoIP environment

Unanswered Question
Sep 29th, 2010

Some ITSPs are offering hosted VoIP services where customer just need to plug&play cheap SIP phones to their
internal network behind firewall, call control etc will be located in the
cloud. Customer does not have to configuration anything special on their firewall. I am
wondering typically how the call works, without the firewall being SIP aware,  how does internal SIP phone tell
other party where to send media to? using STUN to find its external IP/
media port number? use TURN to relay media? also, how does internal phone call work? will
the media be hair-pined? ASA won't allow such traffic.

I have this problem too.
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It does not work well.  I tried vocalocity and cancelled after 30 minutes on the phone.  I am not the most advanced network or phone person, but I have to read enough to learn more.  Essentially what will happen when using cheap phones, they will work like an internet appliance, without VLANS and other QOS settings, your firewall won't be able to tell the difference between a phone call and general web browsing, so if downloading large files, your quality could suffer.  I have a client who still uses it, and his office phone is worse than his cell phone, anytime his receptionist downloads something and he is on the phone the call breaks up or drops all together.  Unless you like poor quality phone calls, worse than cell phones, I wouldn't waste your time with hosted VOIP.  Or if you don't want to invest in IP Telephony equipment, at least invest in a good cisco ASA5505 and set up VLANs and tagging, so you can assign high priority to VOIP packets. 

Then you are at the mercy of your ISP, if your ISP is not your ITSP then your voice packets will not get priority after it leaves your network and goes to the public network, if you don't have dedicated internet, I would stay away from hosted VOIP services, most DSL can cable connection have 60-75ms of latency, which will cause poor quality, and have no guarantee for latency.  So if you want hosted VOIP get it from your ISP and it should work, it it does not come from ISP you are rolling the dice.

Good Luck.

jiangu Wed, 09/29/2010 - 22:20

Thanks for your time, but QoS is not really what I have problem understanding, I am more interested to understand the call signaling and media flow for SIP phones behind a SIP un-aware firewall/NAT device.


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