Dear All,
I have a FS production deployment:
- A FS server behind a NAT network
- User can register and make calls via internet.
It have worked well more than 2 years, but it took problem now:
Assume I have following cases:
1. Phone number in PSTN call to Extension in my branch
2. Extensions from another branch call to Extension in my branch (via sip trunk)
3. An extension in my branch call to another extension in my branch
4. An extension in my branch call to a phone number in PSTN
5. An extension in my branch call to a an extension in another branch (via sip trunk)
These are results:
Cases 1. 2. and 3.: Caller heard well but sometime Calleed can't heard from caller. If i wait for a longer time (may be 1 minutes, 2 minutes...) Calleed might heard from Caller.
Cases 4.5: Both Caller and Calleed heard well.
I have captured PCAP file on FS server, I saw that there were no different between cases 1,2,3 and case 4,5. I have seen many packets that FS server send to calleed IP.
There are only one change to my branch network. Network administrator make a NAT (80 and 8000) for camera service at public interface to a private IP in my branch network.
My question is: why Calleed in case 1, 2, 3 can not heard from caller ?
Another question: FS use RTP for carrying out voice/video but I can only see about 11 RTP packets (for one direction), but there are too many UDP packets in call session. I have attached the picture for that.
In the picture:
Thanks in advance for your help.
--
Best regards,
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EM NGUYEN
Website:
http://freeswitch.vn Mobile: +84 949669075
Yahoo & Skype: nghiaembt
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