Audio delayed, broken, and very slow

[richardnk]

I have just installe V10 on a Window XP SP3, New HP with Intel Celeron (Yuk).
I am using Grandstream BT200 phones, with CallCentric as my provider.

I have tried to make several call and everything seems to be working, however, the audio take about 4 or 5 seconds to get to the other end and when it does it is very broken up and at about half speed.

All internal extensions seem to work well. These effects happen when call out of our network, or into our network using our CallCentric provided toll free number.

I will attach the logs of a couple of the phone calls. If you need any further information please feel free to contact me here or at: [email protected]

Thanks,
Richard

PS. Call number 7 is on that was made from outside of the into the network
Call number 6, i believe was calling out of the network



23:20:47.587 [CM306003]: SIP IPort mapping (12.183.34.2:2093) resolved by STUN server 216.93.246.16:3478 differs from the one (12.183.34.2:61036 resolved by STUN server 178.238.134.190
23:20:47.587 [CM306003]: SIP IPort mapping (12.183.34.2:26348) resolved by STUN server 96.9.132.83:3478 differs from the one (12.183.34.2:61036 resolved by STUN server 178.238.134.190
23:20:47.415 [CM306003]: SIP IPort mapping (12.183.34.2:26348) resolved by STUN server 96.9.132.83:3478 differs from the one (12.183.34.2:2093 resolved by STUN server 216.93.246.16
23:20:47.196 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 96.9.132.83:3478 over Transport 192.168.10.69:5060
23:00:50.587 [CM306003]: SIP IPort mapping (12.183.34.2:8354) resolved by STUN server 216.93.246.16:3478 differs from the one (12.183.34.2:49588 resolved by STUN server 178.238.134.190
23:00:50.587 [CM306003]: SIP IPort mapping (12.183.34.2:59903) resolved by STUN server 96.9.132.83:3478 differs from the one (12.183.34.2:49588 resolved by STUN server 178.238.134.190
23:00:50.415 [CM306003]: SIP IPort mapping (12.183.34.2:59903) resolved by STUN server 96.9.132.83:3478 differs from the one (12.183.34.2:8354 resolved by STUN server 216.93.246.16
23:00:50.196 [CM506004]: STUN request to STUN server 96.9.132.83:3478 has timed out; used Transport: 192.168.10.69:5060
23:00:47.181 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 96.9.132.83:3478 over Transport 192.168.10.69:5060
22:43:45.774 [CM503008]: Call(7): Call is terminated
22:43:37.681 [CM503007]: Call(7): Device joined: sip:[email protected]:5060
22:43:37.681 [CM503007]: Call(7): Device joined: sip:[email protected]:5060;transport=udp
22:43:32.118 [CM505003]: Provider:[CallCentric - US] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]
22:43:32.118 [CM503002]: Call(7): Alerting sip:[email protected]:5060
22:43:30.321 [CM503025]: Call(7): Calling VoIPline:15756872429@(Ln.10000@CallCentric - US)@[Dev:sip:[email protected]:5060]
22:43:30.149 [CM503004]: Call(7): Route 1: VoIPline:15756872429@(Ln.10000@CallCentric - US)@[Dev:sip:[email protected]:5060]
22:43:30.149 [CM503010]: Making route(s) to <sip:[email protected]>
22:43:30.149 [CM505001]: Ext.202: Device info: Device Identified: [Man: Yealink;Mod: T20;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, no-recvonly] UserAgent: [Grandstream BT200 1.1.6.32] PBX contact: [sip:[email protected]:5060]
22:43:30.149 [CM503001]: Call(7): Incoming call from Ext.202 to <sip:[email protected]>
22:40:47.337 [CM306003]: SIP IPort mapping (12.183.34.2:35328) resolved by STUN server 216.93.246.16:3478 differs from the one (12.183.34.2:19930 resolved by STUN server 178.238.134.190
22:40:47.337 [CM306003]: SIP IPort mapping (12.183.34.2:39192) resolved by STUN server 96.9.132.83:3478 differs from the one (12.183.34.2:19930 resolved by STUN server 178.238.134.190
22:40:47.165 [CM306003]: SIP IPort mapping (12.183.34.2:39192) resolved by STUN server 96.9.132.83:3478 differs from the one (12.183.34.2:35328 resolved by STUN server 216.93.246.16
22:40:46.946 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 96.9.132.83:3478 over Transport 192.168.10.69:5060
22:33:44.322 [CM504004]: Registration succeeded for: 10000@CallCentric - US
22:33:44.025 [CM504003]: Sent registration request for 10000@CallCentric - US
22:33:41.540 [CM504004]: Registration succeeded for: 10000@CallCentric - US
22:33:41.150 [CM504003]: Sent registration request for 10000@CallCentric - US
22:31:00.806 [CM503008]: Call(6): Call is terminated
22:30:49.259 [MS105000] C:6.2: No RTP packets were received:remoteAddr=127.0.0.1:40614,extAddr=0.0.0.0:0,localAddr=127.0.0.1:7010
22:30:48.228 [CM503007]: Call(6): Device joined: sip:[email protected]:40600;rinstance=1e0f7edc7defb11e
22:30:48.212 [CM505001]: Ext.999: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Voice Mail Menu;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] PBX contact: [sip:[email protected]:5060]
22:30:48.212 [CM503002]: Call(6): Alerting sip:[email protected]:40600;rinstance=1e0f7edc7defb11e
22:30:48.072 [CM503025]: Call(6): Calling Ext:Ext.999@[Dev:sip:[email protected]:40600;rinstance=1e0f7edc7defb11e]
22:30:48.025 [CM503005]: Call(6): Forwarding: Ext:Ext.999@[Dev:sip:[email protected]:40600;rinstance=1e0f7edc7defb11e]
22:30:48.025 [CM503016]: Call(6): Attempt to reach <sip:[email protected]:5060> failed. Reason: Not Registered
22:30:48.025 [CM503017]: Call(6): Target is not registered: Ext.200
22:30:48.025 [CM503010]: Making route(s) to <sip:[email protected]:5060>
22:30:48.025 [CM505003]: Provider:[CallCentric - US] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]
22:30:47.165 [MS211000] C:6.1: 204.11.192.37:52794 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
22:30:42.900 [CM503007]: Call(6): Device joined: sip:[email protected]:40600;rinstance=1e0f7edc7defb11e
22:30:42.900 [CM503007]: Call(6): Device joined: sip:[email protected]:5060
22:30:42.884 [CM505001]: Ext.999: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Voice Mail Menu;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] PBX contact: [sip:[email protected]:5060]
22:30:42.884 [CM503002]: Call(6): Alerting sip:[email protected]:40600;rinstance=1e0f7edc7defb11e
22:30:42.744 [CM503025]: Call(6): Calling Ext:Ext.999@[Dev:sip:[email protected]:40600;rinstance=1e0f7edc7defb11e]
22:30:42.728 [CM503005]: Call(6): Forwarding: Ext:Ext.999@[Dev:sip:[email protected]:40600;rinstance=1e0f7edc7defb11e]
22:30:42.728 [CM503016]: Call(6): Attempt to reach <sip:[email protected]:5060> failed. Reason: Not Registered
22:30:42.728 [CM503017]: Call(6): Target is not registered: Ext.200
22:30:42.728 [CM503010]: Making route(s) to <sip:[email protected]:5060>
22:30:42.712 [CM505003]: Provider:[CallCentric - US] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]
22:30:42.712 [CM503001]: Call(6): Incoming call from 15756872429@(Ln.10000@CallCentric - US) to <sip:[email protected]:5060>
22:30:42.587 [CM503012]: Inbound any hours rule (unnamed) for 10000 forwards to DN:200
22:30:31.509 [CM503008]: Call(5): Call is terminated

 

[eagle2]

Have you run 3CX firewall checker ? What is the application exit code ? It seems like you are having some problems with NAT / firewall.

Can you apply some kind of QoS (quality of service) in your network router ? It seems like you having network problems with Internet also.

Regards

 

[richardnk]

eagle2,

Thanks for your responce.

The application exit code on the firewall checker is: 53

I am looking into firewall issues.

Thanks again,
Richard

 

[richardnk]

By the way, I am using a MonoWall for a firewall.

Everything seem to be set properly....never the less the problem remains.

Thanks Again,
Richard

 

[eagle2]

Hi Richard,

Unfortunately not, at least for 3CX PhoneSystem.
You must have application exit code = 0 from Firewall Checker for reliable operation.

You need to forward (a kind of destination NAT) TCP ports 5060 and 5090 and UDP ports 5060, 5090, 9000 - 9049 to local address of 3CX server without SIP ALG been active (on some routers).

STUN errors in your log may be are indication for incompatibility with SIP ALG (otherwise try disabling STUN in 3CX server).

Regards,
Orlin

 

[willow]

what type of internet connection do you have and can you run a few tests at speedtest.net and pingtest.net and post the results.

 

[richardnk]

eagle2, willow,

Thanks again for your help.

I work nights at an astronomical observatory in south central New Mexico and had the last two nights off, so sorry for the delay in getting back to you.

eagle2: all the mentioned ports are open and forwarded. I'm not sure I under stand what you were trying to say here: " local address of 3CX server without SIP ALG been active (on some routers). ", however, I have tried operations with and without the STUN server, with the same results.

willow, I am on a LAN here, 4 T1 lines. Behind a router, of course and a m0n0wall firewall. I have operated a Trixbox from this environment for the last 3 years without any problems....well....for the most part.

The speed test shows: down 5.36 mbps, up 1.85 mbps. The ping test shows: packet loss 0%, ping 81 ms, jitter 2 ms. This is to Los Angeles from a remote location 32 miles up into the mountains east of Alamogordo, NM. Not bad actually. Of course these numbers can vary depending on which server I test to.

Once again, the SIP protocol seems to be working fine. I can call out or in and the connections are made. I can here audio in the communication, however, it is delayed by a few seconds, very jittery, and at about half speed. The problem seems to be in the RTP packets, perhaps.

Thanks again for your time. If there is anything else I can provide to help, please ask.

Thanks,
Richard

 

[eagle2]

Hi Richard,

read this post: http://www.3cx.com/blog/voip-howto/firewall-nat-pat-stun/

Some routers (like Cisco) support SIP ALG, which translates public addressesorts to internal addressesorts, changing not only address fields, but also SIP headers. This kind 'SIP NAT/PAT' is required for proper operation in NAT environment. The problem is some routers are not doing this correctly, so 3CX generally recommends switching off SIP ALG (Application Level Gateway functionality) and using STUN instead of this to resolve your public address and substitute it into SIP messages. It you have both STUN and SIP ALG active, you may run into similar problems like yours. To disable STUN read: http://www.3cx.com/blog/voip-howto/stun-resolution/ (if STUN is the problem, Trixbox should be working without STUN).

That's what I meant. Making SIP working behind a NAT router sometimes could be complicated. It looks like you have some firewall / NAT problem. You'll definitely need a kind of QoS (Quality of Service) to solve other issues (possible congestion, etc.) with prioritizing VoIP traffic. Unfortunately I don't have experience with m0n0wall.

Regards,
Orlin.

 

[Eurylink]

If you are using Monowall you have to disable queuing and setup Monowall to do static port mapping (by default it do dynamic port mapping). Maybe you'll be better to use Pfsense instead of Monowall in Voip application environment because in 2.0 version Pf is able to do static port mapping and there is a simple way to enable traffic shaping in a voip compatible way.

 

[eagle2]

If you eventually think of changing router, consider also Mikrotik - the RB750G is a very good price / performing model, lots of functionality, VPN, VLAN support, easy setup, works fine with 3CX phonesystem. I could provide remote assistance for setup.

Regards