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Joined: Jul 2005
Posts: 210
OP Offline
Joined: Jul 2005
Posts: 210
ON my SV8500 I have mostly DT700 IP phones,but have a few SP350 softphones and have many 3rd party analog gateways (Sangoma Vega 5000). I have plenty of proper licenses for each end-points.
Recently it was reported to me that when a user of an analog phone connected to the analog gateway calls an SP350 softphone, there is only 1-way audio. The analog-phone user can hear the softphone user, but the softphone user can't hear the analog phone. No matter who initiates the calls.
So I started testing this out under different circumstances, and it always boils down to just that pair: 3rd party SIP and SP350.

-3rd party SIP to DT700 GOOD both ways
-3rd party SIP to 3rd part SIP GOOD both ways
-3rd party SIP to SP350 FAIL
-DT700 to SP350 GOOD both ways

While we have many different subnets and VLANs, I ruled out access list problems and packet block by:
- for the sake of test, removed ALL blocking between all networks. The problem persisted.
- Tried to put the 3rd part SIP gateway on the same subnet as the SP350. Problem persisted.

Lastly, I installed Wireshark on the PC that hosts one of the SP350 phones. It captured voice packets going both ways. Wireshark has a feature to decode VOIP calls from its captured packets. I did that. It found that call and played it back to me nicely with 2-way audio!!! So really, SP350 is somehow just refusing to play the sound of the 3rd party SIP phone. But it can't be an audio setting on the PC or the SP350 itself since it plays nicely the sound of other SP350 callers or other DT700 callers.

So what on earth is going on??? I'm using SP350 version 5.2.0 (I think) with HP Thinclients running Win 7, but the problem existed on a Win XP machine and SP350 version 4.something.

I compared the voice packets coming to the SP350 from the DT700 and from the 3rd party SIP device, and found the below differences:
-DIFFSERV value (0x00 for the 3rd part SIP and 0x8b for the DT700)
-UDP port number is 3462 for the DT700 and 10000 for the 3rd party SIP device.
-the packet length is different
- They show the same codec: G711u

I then reconfigured the 3rd party device to use the same UDP port and same DIFFSERV value as the DT700, and STILL NO AUDIO!
I contacted my vendor's tech support who is pointing me towards playing with the LOC-ID settings in the AIVCL/N commands. NEC recommends a G711 with payload of 20ms of G729 with payload of 40ms setting. I tried those but still nothing.

The install manual of the SP350 says that Win 7 only supports 4 levels of TOS setting for the SP350, and the PBX has to be configured accordingly. However, it doesn't say how to find out what is the TOS setting in my Win 7 computer for the SP350, thus I have no idea how to match that in the PBX.

Anyways. This isn't going anywhere. Oh by the way, I then tried a Grandstream analog SIP gateway just for a test and the exact same audio problem exists.

Any great ideas?

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Joined: Jul 2005
Posts: 210
OP Offline
Joined: Jul 2005
Posts: 210
This is now solved. The problem was that by default, NEC uses G711 codec with 40ms payload size between its own IP terminals. So when a call was happening between the NEC softphone (or any other NEC terminal for that matter) and the 3rd party SIP device, the NEC was trying to use the above codec, while the 3rd party SIP device was unable to play that. My 3rd party SIP device was a Sangoma Vega 5000 analog gateway. The Sangoma ignored what NEC was trying to do, and was playing its own codec, which was G711 with 10ms. For one reason or another, all NEC terminals understood that, except for the NEC Softphone. I mean I went as far as putting wireshark on the computer that had the NEC softphone on it, and recorded the conversation. And while the softphone user couldn't hear what was coming from the Sangoma user, wireshark recorded a nice 2-way audio on that VERY same computer.
While working on this, I tried a setting on the Sangoma to observe and then honor the codec request of the NEC system, and play audio on G711 with 40ms payload. But that codec doesn't exist on the Sangoma, so all audio failed at that point.
I worked with my vendor, who quoted and NEC Knowledge Base article about a known potential audio problem with the NEC SP350 softphone. In that they say, if there is 1-way audio experienced while using the SP350, do the following:
Create "locations" within the system by the use of ALOCL command. Basically, you would put a set of phones (say in your local office) into location 1, while the phones in a remote office would be put into location 2.
Then use the AIVCL/N command to define codec and other communication preferences for calls happening between those 2 locations, and NEC suggested that calls involving the softphones should be set to G711 with 20ms payload or G729a with 40ms.

I didn't have different office locations, but the Sangoma and the softphones were on different network segments on my LAN, and I defined 2 different locations. One for the network segment where the Sangoma was, and another one for the network where the softphones were.
Then I used the AIVCL/N to specify codec G711 with 20ms for calls between these locations. Then on the Sangoma I forced its codec to G711 with 20ms, and VOILAH!!! There was 2-way audio.
Many thanks to both NEC and Sangoma tech support. I hope somebody will benefit from this write-up in the future.

Moderated by  ttech 

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