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

[Dovid Bender]

Ok. Then Asterisk must see the external IP and think that the device is not
behind NAT. A way of "tricking" Asterisk is to set up the customers external
IP as localnet on Asterisk and see if that fixes it. If this is the case
time for a new router....


-----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 16:15
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] No audio on some routers with PAP2T

yes net is set to yes on both

On Mon, Jan 30, 2012 at 4:08 PM, Dovid Bender <os-list at dovid.net> wrote:
> 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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