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--- RE: One-Way Audio (http://www1.pbxes.com/forum/threadid.php?threadid=1220440019)

Audio Recording

Hi,
I have been recording calls in order to get to the bottom of some agents complaints about failures.
I noted several inbound calls yesterday where the recordings have clear audio from both parties yet the caller cannot hear the agent (although the agent can hear the caller).
How does this happen ?
Ron

RE: One-Way Audio

Are the agents SIP User Agents behind a firewall or on Public IPs?

RE: One-Way Audio

Behind a Netgear router.
I have fixed the local IPs, separate SIP ports, and port forwarded SIP to appropriate ATA.
Should I now forward RTP also ?.
I assume I can set a unique range for each ATA.

I have used pingplotter and identified spikes of packet loss when various PCs etc have accessed particular internet sites, so I have also purchased a router with QOS (Draytek Vigor) which I hope to install tomorrow to see if it makes a difference.
Regards
Ron

RE: One-Way Audio

I hope the Draytek router you got has a couple of FXS ports on it too. I can assure you will get perfect service out of them, if you set them up properly.

Regarding the rest of the ATAs behind the Draytek, setup each of them on a unique RTP port range, and forward the RTP ports through the router to their private IPs.

Reserve the signaling port 5060 for the ATA of the Draytek, since it doesn't seem to work on any other port. Both FXS can coexist on 5060.

RE: One-Way Audio

No, I got a 2820n without FXS.
My ATAs are SPA units, and I just received a 941 phone to trial (appears much the same as the SPA ATAs in the setup).
I'm not quite sure what RTP port range to set and to forward for each (7) ATA. How many RTP ports does an ATA require ?
I note they are all currently at default 16384 - 16482.
Regards
Ron

RE: One-Way Audio

Hey Ron,

Assign a range of about 10 RTP ports for each ATA, and that will be enough.

Be aware the SPA-9xx phones sometimes face an issue with sending DTMF digits, since you can't turn off DTMF generation of either RFC2833 or SIP Info.

The ATAs don't have that issue, since there is an extra setting controlling DTMF generation for both RFC2833 and SIP Info. Linksys is aware of the issue but haven't fixed it yet.

RE: One-Way Audio

Thanks for the quick reply.
I have tried to keep order in ATA settings, ie ext 101 uses SIP 5061, so I will assign it 10 RTP ports starting 16384.
Ext 102 is SIP 5062 and hence RTP 16394 up.
I will set the range in each ATA and then port forward.
Does that sound ok ?
I will check the 941 operation carefully when I get round to trying it.
Just need to nail down the reliability of the basic system.
Ron

RE: One-Way Audio

What model are your SPAs, and how many of them do you have in a single location?

Keep in mind their default signaling range is from 5060 to 5080.

RE: One-Way Audio

7 Office users and 2 homeworkers.
The office has 3 * 3102, 1 * PAP2-na, 2 * 1001.
Homeworkers are 2 * 1001.

The FXOs of the 3102s are also connected and working, but will be disabled when we can get the published number moved to an internet trunk.

The overall number of ATAs will not increase.

RE: One-Way Audio

Well all of your ATAs require 2 SIP ports each, so in the office you need 12 unique ports.

RE: One-Way Audio

Hi,
Well very impressed with the Draytek router, not one stuttering call since I put it in last week. I found the QOS a bit confusing, but an easy guide off the net worked a treat. The built in diagnostics are informative.
Haven't forwarded RTP yet as I am still unsure of the multitude of settings on the SPA units, however no reports of one way audio either !
My backup outgoing 3102 FXOs also now work (coincidence ?) with 'dynamic' setting instead of having to point at the dyndns:port.
The lesson seems to be start with a good router !!
Ron

RE: One-Way Audio

... or move on to the Holy Grail of Public IPs for every SIP UA.

RE: One-Way Audio

One issue I still have:
An extension in the ring group not ringing, then starting ringing after the call has already been picked up on another extension.
Its generally, but not always, the same extension, occasionally several. The log shows all the extensions called, but sometimes an extension not showing as ringing until a few (3) seconds later (after the call has been picked up).
This has happened before (and is the reason I first forwarded sip ports), indeed I raised it in this forum, it went away for a while and then started again on the weekend.
New ring group created without effect.
Ron

RE: One-Way Audio

A couple of questions regarding this issue:

• Are all the extensions which have the non-ringing issue in the same LAN segment?
• Have you tried a Submit & Start in the Personal Data section?

RE: One-Way Audio

Submit and restart several times.
There seems to be just 2 extensions that display this behaviour, both are 3102s wired to the router. The third wired 3102 doesn't have an extension on it. All other extensions are wireless (and haven't done this false ring).
The log clearly shows all extensions called together, then all but one ringing, then an extension answers and the rest hung up. Then several seconds later phantom ring on one of the wired 3102s which is picked up and sip 481 generated.
Maybe I should go all wireless ?
Ron

RE: One-Way Audio

Hey Ron,

Can you please clarify what you mean by wireless? Are all the ATAs connected to the LAN via WiFi bridges? Are the FXS ports of the PAP2 and the SPA-1001s connected to cordless phones?

Also when you wrote: "... then an extension answers and the rest hung up", did you mean stop ringing instead?

Have you setup an ethernet packet trace to capture the SIP signaling in your LAN?

RE: One-Way Audio

Hi,
My wireless extensions are just those ATAs that connect to the Draytek router via its wireless (I use small linksys wireless devices) rather than straight into one of the 4 ethernet ports.
All ATAs have a standard telephone plugged in (no cordless etc).
All references are to the PBXes log, ie extensions called, extensions ringing, extensions hung up.
In the log normally all extensions are called at the same time, then ringing at the same time, then hung up together at the same time (except for the answered one). The call then continues normally.
sometimes one extension is missing from the 'ringing' log entry (and physically isn't ringing). It then rings a few seconds later (usually after the call has already been picked up), and continues to ring until picked up and put down again.
Its just like the ring 'command' has been delayed a few seconds and not received by the extension until after all the other phones have rung and unused ones hung up by PBXes.
This has happened before (before router change) and seemed to go away again, but now its back !
No other problems.
Ron

RE: No concurrent ringing

Hey Ron,

If you need to troubleshoot this issue further, setup an ethernet packet trace by connecting all the ATAs on a hub, while you run a packet trace utility on a PC. If you wish to do this by keeping the current setup, install a Syslog server on a PC in your LAN, and instruct all the ATAs to send their debugging info to it.

This way you will be able to see when the ringing message arrives to each ATA.

RE: One-Way Audio

Hi,
Not been able to set up a packet trace yet, but more info on what I observe happening.
1) Maybe 20% of incoming calls display the problem, can be spread out or in a bunch.
2) Seemed to always be one of 2 extensions out of total 7 so I deleted and renumbered those 2 extensions and the problem just shifted to another extension that had never done it before.
3) If I leave the incoming call unanswered then the extension will start ringing 8 seconds after the others and the call can be picked up on it normally.
4) The PBXes log shows the extension missing from the 'ringing' entries. If the call is picked up on another extension then a SIP 481 is generated and after 8 seconds a SIP reinvite which appears as the 'phantom' call.
No other problems, call quality is fine, no one way audios or stutter even with 4 or 5 simultaneous G726 calls.
Ron

RE: One-Way Audio

Hey Ron,

This is a very strange issue, but there is a definite method to determine whether all the Invites arrive at your network, or if there are delays in getting there.

In order to setup this test environment, all the ATAs should be connected to an ethernet hub (not a switch) with a PC running a packet capture application, such as Wireshark.

We will then look at the trace files, compare them to the traces from the SIP Proxy side, and try to figure out where the issue lies.

We can try to walk you though the packet trace procedure, so let us know if you need help.