【Phenomenon】
The IP telephone call from Site1 to Site2 is not working.
【Problem analysis】
Step 1: Capture packets on the AR. The ACL used for filtering is 192.168.80.20 and 192.168.1.2. No packet is captured on the AR.
Step 2: Make sure that the test call is from SITE1 to SITE2. Therefore, it is suspected that packets are lost on the link between U1911 and the USG.
Step 3: The INVITE packet from 192.168.80.20 to 192.168.1.2 is captured on the internal network port of the firewall, but no packet is returned.
Step 4: No exception is found on the firewall.
Step 5: Capture packets on the U1911 and U1960 without considering the USG and AR. The U1960 receives the INVITE message sent from 192.168.1.254 to 192.168.1.2.
Step 6: According to the analysis of the entire network, it is preliminarily determined that the AR sends the INVITE message to the source NAT. Convert 192.168.80.20 to 192.168.1.254.
Step 7: Check the configurations on the AR; SIP ALG is available. NAT is performed on the SIP packets. Delete the SIP ALG configuration and clear the related flow table. The test service is normal.
【Root cause】
The customer configured 'nat alg sip enable' on the AR router in France; this command has done the NAT for the SIP packets sent by IP telephone. After undo nat alg sip enable, the issue is resolved.
【Suggestion】
In the case of follow-up questions like this, capture the packets on the two SIP servers to check whether the packets are lost or error packets are received. Then, check the network devices based on the error causes.
