[OpenSIPS-Users] No audio on some routers with PAP2T

[Dovid Bender]

Shnuer,

The issue is that when the packet comes from behind the Baudtech Asterisk
see's an external IP and there for thinks there is no NAT and is looking for
the RTP on the ports stated. When you send from the Linksys it see's a local
IP and knows there is NAT. Do you have nat=yes in sip.conf in both the
general section and for the IP of OpenSIpS?

Rergards,

dOvid


-----Original Message-----
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Schneur Rosenberg
Sent: Monday, January 30, 2012 15:07
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] No audio on some routers with PAP2T

Hi, I have a openSIPS server setup to do registration and load
balancing between 2 Asterisk servers, the Asterisk servers do
everything besides registration and they are load balanced by the
openSIPS servers, incoming calls hit the openSIPS server which sends
it to the Asterisk server and if it needs to go to a local phone it
sends it back to openSIPS where the phone is registered to, outgoing
calls get sent to Asterisk via load balancing and asterisk completes
the call.

I have a problem with some ata's (in my case pap2t) that when its
behind certain routers (in my case a Baudtech) there is no audio, when
I try a different router it does work, also when I try a different ata
like a spa2102 it does work, also when I connect the pap2t directly to
the asterisk it works fine, NONE of the routers have SIP ALG enabled,
it seems that the nat blocks the audio when the media is from a
different server.

The interesting thing is that the Baudtech router changes the internal
IP's to the external ip, the other router does not, does that mean
that there is some kind of ALG built into the Baudtech router? even if
it does how come the Asterisk server handles the audio fine while the
openSIPS breaks the audio

here is a trace of the initial INVITE from the Baudtech (the
problematic one) as you can see the ip at the Via and Contact and in
the c tag in the RTP have been replaced by the router

U 46.116.60.131:5060 -> 64.69.33.43:5060
INVITE sip:18005558355 at sip.myserver.com SIP/2.0.
Via: SIP/2.0/UDP 46.116.60.131:5060;branch=z9hG4bK-278a1ace;rport.
From: <sip:customer1191 at sip.myserver.com>;tag=aea4a93b350d6746o0.
To: <sip:18005558355 at sip.myserver.com>.
Call-ID: d9e5241f-e0a89b2 at 10.0.0.3.
CSeq: 101 INVITE.
Max-Forwards: 70.
Contact: <sip:customer1191 at 46.116.60.131:5060>.
Expires: 240.
User-Agent: Linksys/PAP2T-5.1.6(LS).
Content-Length: 442.
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
Supported: x-sipura, replaces.
Content-Type: application/sdp.
.
v=0.
o=- 50374 50374 IN IP4 46.116.60.131.
s=-.
c=IN IP4 46.116.60.131.
t=0 0.
m=audio 16476 RTP/AVP 0 2 4 8 18 96 97 98 100 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729a/8000.
a=rtpmap:96 G726-40/8000.
a=rtpmap:97 G726-24/8000.
a=rtpmap:98 G726-16/8000.
a=rtpmap:100 NSE/8000.
a=fmtp:100 192-193.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendrecv.

Here is the same invite when send from the DLINK router (here audio is fine)

U 85.250.89.78:5060 -> 64.69.33.43:5060
INVITE sip:18005558355 at sip.myserver.com SIP/2.0.
Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bK-65cff5ef;rport.
From: <sip:customer1191 at sip.myserver.com>;tag=2ab40bee91703297o0.
To: <sip:18005558355 at sip.myserver.com>.
Call-ID: 93503bfa-3a0ce1fb at 192.168.2.100.
CSeq: 101 INVITE.
Max-Forwards: 70.
Contact: <sip:customer1191 at 192.168.2.100:5060>.
Expires: 240.
User-Agent: Linksys/PAP2T-5.1.6(LS).
Content-Length: 442.
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
Supported: x-sipura, replaces.
Content-Type: application/sdp.
.
v=0.
o=- 19246 19246 IN IP4 192.168.2.100.
s=-.
c=IN IP4 192.168.2.100.
t=0 0.
m=audio 16438 RTP/AVP 0 2 4 8 18 96 97 98 100 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729a/8000.
a=rtpmap:96 G726-40/8000.
a=rtpmap:97 G726-24/8000.
a=rtpmap:98 G726-16/8000.
a=rtpmap:100 NSE/8000.
a=fmtp:100 192-193.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendrecv.

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