Random One Way speech

[davea2]

Hi

I've been scratching my head on this one.

recently installed server edition with Primary and Secondary in different locations and about 7 remote sites hanging off each all over MPLS with breakout for sip trunks from both main sites.
All works fine apart from random sporadic one way speech on an inbound call from the sip trunks.

Only happens to a few and if the caller dials straight back in chances are the call will be fine.

The MPLS is new too and they have deployed BGP for some strange reason. Firewalls are Sonicwall unfortunately.

It's the random nature of the problem that's puzzling, any ideas?

Cheers!

 

[pkdev]

Try to disable SIP ALG on firewalls

Do you have set up direct media path ?

What phone do you use (we have a lot of issue with J1XX series)

 

[SamIPOfficeSupport]

Hi there Dave,

Intermittent one-way audio is a fun one... Do you know if it ever occurs on internal-only (extension to extension) calls? If it's isoalted to calls that involve the SIP trunks, then please send us a screenshot of the SIP settings in the IP Office configuration, and also the System>Voip tab so we can see the codecs that are allowed at the system level.

 

[IPGuru]

can you get a wire shark trace before the circuit enters the IPO
this should enable you to see if the speech is even being delivered to the IPO (if not then it is a provider issue & nothing you can do).



Do things on the cheap & it will cost you dear

 

[SamIPOfficeSupport]

I agree with IPGuru, getting a packet capture would prove if the speech path is being dropped externally. You can listen to the RTP stream from wireshark too. With that as proof, you can push it back to the SIP provider.

 

[davea2]

Update on the one way speech.
Therer are occaisionally one way speech on internal calls (site to site)
One site has no audio at all on internal calls within the same branch or on site to site but calls on the SIP trunks are fine.

External calls only seem to be a problem inbound

Also there seems to be random call drops ate about 15 minutes.
I am told that the calkl does not actually drop, the speech disappears but the handsets looks like it's still connected

I have had similar problems to these before and they generally are the firewall, but I am told all is good with those...


Struggling to get PCAPs at present

 

[davea2]

I should add the system is 11.0.0.0 and all handsets are J169 or J179

 

[SamIPOfficeSupport]

Does this only occur on transferred calls? If so, the fix for that is to turn on SIP refer in the IP Office SIP line settings.

If it is not transfer-related, but is specific to the 15-minute duration mark, I think you need to do some more testing with two phones on the same network. I'd test with settings as-is (internal call from phone A to phone B)... see if you can get it to happen. Then, do it again after you uncheck Allow Direct Media Path for the two SIP extensions. Might be a good time to make change the system level codecs so that only G.729 and 711 are allowed (or match the codecs being used by the SIP provider, which usually seem to be 729 and 711)

Also, when you get the speech path to drop, go ahead and try putting the call on hold, and then picking it back up to see if speech path is restored.

 

[davea2]

Hi Sam

I had already treid reducing the allowed codec to G.711 only in case it was a codec mismatch.
I have asked the end user to try the hold/unhold too - no reply yet.

TRhe one way speech is on direct dialled calls site to site. and direct inbound.
Some calls go via an auto attendant but not ure this changes the end result in any way

I was considering upgrading to SP4 and the handsets to 4.0?

 

[SamIPOfficeSupport]

Definitely upgrade to latest SP before spending too much time.. just in case.

Are you sure that the SIP is using 711? Usually it's set to 729, then 711. SO ideally' you'd want both allowed on the IP Office, in that order. Also, what do you mean by site to site call? What do you mean by direct inbound?

 

[davea2]

Yes, all is set to G711 only, even the ITSP.

By direct site to site they have a primary amd a secondary with about 7 sites with remote phones hanging off each all over MPLS, routed.
Direct inbound is sip trunk direct to a phone or HG. It could be to a phone on the site where the server is or to one of the remote sites.

95% of calls seem OK...