I'm fighting with Cisco 7960 running 8.6 SIP software. I've tried to get that phone working behind NAT. No success at all. I got SIP messages working, I could place and receive a call but no RTP stream ever begins, even one packet.
Anyway, I have notticed that I got no dial tone when I take handset off hook. I suppose that somewhere software upgrade process has failed or spoiled something inside the phone (DSP registers? other hardware registers?).
Should the dial tone be there after registering on SIP Proxies? Is dial tone a neccessary thing? I only hear audible buzz/background noise.
The listen port for SIP messages is configurable. When the NAT enable flag is used in conjunction with the VoIP Control Port, the packets are sourced from this port rather than from an ampherol port. See the voip_control_port parameter descriptions in the section, "Modifying the Default SIP Configuration File" in Chapter 3, "Managing Cisco SIP IP Phones," at the following URL:
Thanks for the document! Unfortunately I have those options enabled and correctly set. I have tried nat_received_processing parameter too. I've tried setting outbound_proxy to address of my router where SIP Express Router software runs. No success. I've checked all the configuration combinations step by step, one by one to eliminate conflicts or misconfiguration. I have forwarded all required ports (control port, media ports). I even put the phone into DMZ to get all of them available. I have my VoIP lines registered on corresponding proxies , I could place and receive a call but still no voice. I've dumped some packets and I see no RTP stream at all even if SIP negotiation seemed ok and no "private" adresses appear inside.
The only option available for me now is to put the phone directly into internet with public address and no firewalls and NAT at all....
I would check the device that is doing NAT / firewall. If this one does not dynamically open the UDP ports, the RTP stream has no chances to pass. If you disable VAD on the phone you should be able to observe/capture RTP up to the NAT device.
I tried two NATing devices: Cisco 1605 router and Linksys WRT54GL. I set both to forward traffic directed to media ports (16000-20000) onto my phone behind NAT. That should be enough. I put my phone into DMZ to make all ports available for it. I tried ALG mechanism on Cisco 1600 with and without nat option enabled on the phone. No success at all.
I see no RTP traffic trying to get out of the phone even with VAD disabled (and still no dial tone). I used old hub for sniffing traffic between devices.
p.s: it is not 7960G phone, it is 7960 with english hard keys (older version), if that does matter..
indeed the RTP stream shoulod begin immediately after the call setup succeed. Perphaps the RTCP mechanism is't working, however if the phone works when you connect directly to the iternet, you can be sure the problem is cause by NAT / Firewall.
The short answer is that you don't.... That isn't entirely true while at
the same time it kind of is, but for the most part you don't configure
the softkeys. You enable or disable them via TCL. Here is the long
answer. Be sure to read the whole thing or e...
Topology: IP Phone > Switches > Microsoft NPS setup to forward 802.1x
proxy to > ISE 2.1 patch 3 Authentication: EAP-TLS using Cisco MIC SANs
Phone Models 802.1X support? 802.1x flavor Addtl Comment EAP-MD5 EAP-TLS
Cisco 3905 Y Y N Cisco 6911 Y Y N Cisco ...
This document describe how DST changes and how time changes are
implemented in DST. Daylight Saving Time (DST) is the practice of
setting the clocks forward 1 hour from standard time during the summer
months, and back again in the fall, in order to make b...