For my home/personal study I have two routers.
Internet -- Cisco 1841 -- Cisco 1751-V
The 1841 is a ADSL router with NAT, the 1751-V has two FXS and one FXO ports for the connection of PSTN phones. I've configured the sip-ua client (connecting to my ISP) in the 1751-V and that works for 90%. I get the famous one-way-audio problem. After some wiresharking I've found the issue. Multiple c= (connection information) fields are present in the SIP/SDP Invite message my 1751-V is sending. Normally this is not an issue. The NAT SIP service in the 1841 should translate all the entry's within the SIP/SDP Invite message and all would work fine. One entry is for session level and one entry is for media level. But in the 1841 (IOS= 12.4.(20)T) there is a nasty bug (CSCsv43242) which only translates (ip nat service sip) the first entry of an SIP message. Since this bug is present in almost any version available to this router and Cisco has given the priority "6" to this bug, I don't expect a quick solution. The workaround suggested is to prevent multiple c= fields to be send out by the SIP device. Now that's my question, how can I only send out one c= field from the 1700? Since this router is not supported in IOS version 12.4(20)T I can not use sip-profiles to strip the content. Anybody else has an idea? I've tried to modify the c= field with the bind command to a lo0 adapter (with my internet ip on it). But that does not seem to work. Since I'm pretty new to SIP/IPT some help in the right direction would be much appreciated!