Symmetric NAT problem

Hi all,

I am new to asterisk. I unable to hear the voice over internet bcoz both UAs are behind Symmetric NAT Sad, I was hanging with this for past one month. Pls help me to solve this problem? will openSER solve this problem:?:

Thanks in advance

Regards,
Raguraman P

How is your router set up ? What issues do you have ? If OpenSER is behind NAT as well I doubt it will help.

My problem as follws:

Green Line indicates : voice transmitted between my to local machine(I can hear voice)
Red Line indicates: I unable to hear any voice between two machines in the internet.

[Note: Both UA machines are behind NAT. Asterisk Server not behind NAT]

pls help me…

Regards,
Raguraman P

If the end user is behind NAT then the RTP MUST go through the asterisk server. You can not have the rtp go from one user to another when you are behind NAT with Asterisk.

How to make rtp go through asterisk? will it make more traffic on asterisk.

Thanks.
Cheers,
Raguraman P

My sip.conf as follows:

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=yes
directrtpsetup=yes
rtcachefriends=yes
rtupdate=yes

[111]
type=friend
username=111
secret=password
nat=yes
host=dynamic
reinvite=no
canreinvite=yes
qualify=yes
dtmfmode=inband
callerid=111
disallow=all
allow=ulaw
context=from-sip
rtpkeepalive=10

[222]
type=friend
username=2222
secret=password
nat=yes
host=dynamic
reinvite=no
canreinvite=yes
qualify=yes
dtmfmode=inband
callerid=222
disallow=all
allow=ulaw
context=from-sip
rtpkeepalive=10

I have a similar question:

I have users registering to my asterisk box which is on the internet (static IP, no NAT).

If two users are behind the same NAT(they work in the same office) , will the RTP stream still go through my Asterisk box?

sri2talk:
Yes. The RTP needs to go through asterisk.

ragu.nm@gmail.com:
Change your sip.conf to what I have below. Note the nat=yes and canreinvite=no. If you can an sbc you may not need the rtp to go through asterisk. If the rtp goes through asterisk it does put a bit more of a load on asterisk and it will eat up a lot more bandwidth.

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no
directrtpsetup=yes
rtcachefriends=yes
rtupdate=yes
nat=yes


[111]
type=friend
username=111
secret=password
nat=yes
host=dynamic
reinvite=no
canreinvite=no
qualify=yes
dtmfmode=inband
callerid=111
disallow=all
allow=ulaw
context=from-sip
rtpkeepalive=10

[222]
type=friend
username=2222
secret=password
nat=yes
host=dynamic
reinvite=no
canreinvite=no
qualify=yes
dtmfmode=inband
callerid=222
disallow=all
allow=ulaw
context=from-sip
rtpkeepalive=10

Thank u . It’s working fine