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-   -   One way audio but only with some PSTN numbers (http://forum.sipbroker.com/showthread.php?t=4706)

rogerhilary 01-13-2010 07:06 PM

One way audio but only with some PSTN numbers
 
Any help appreciated.
I have a consistent problem with one way audio, outbound from me is fine, but inbound to the PAP2 there is no audio received, however this problem only occurs with some Sipbroker PSTN access numbers.
I am using a PAP2 connected to a Netgear MBR624GU 3G router. The PAP2 is registered to Callcentric. Echo test works OK and I have received a inbound SIP call with good two way audio.
I am not using STUN (syslog reports that I am using a symmetrical NAT)
I have the other line of the PAP2 registered with Sipdiscount and all works well including a UK DID number which allows incming calls from the PSTN.

The problem I experience is that using Sipbrokers PSTN access numbers on the callcentric registered line for some of the numbers there is one way audio whereas for others all is OK with good 2 way audio. I do not understand how this can be as I thought that on a call from a PSTN into a Sipbroker number that Sipbroker would then just pass the call onto Callcentric.
The problem is entirely repeatable
The ones I have tried which al;ways have the one way audio (no audio inbound to the PAP2) are:
All 7 of the listed Australian numbers -(Voxbone and Pennytel)
Both Auckland (NZ) numbers - (DIDWW and Callcentric)
A Boston (USA) number -Callcentric
An Oxford (UK) number - DIDWW

Whereas the following always have good two way audio:
An Oxford (UK) number - VOIPtalk
A Manchester number - VOIPtalk
A London (UK) number - Telappliant

A theme which has emerged is that all the one way audio ones have the male greeting on connecting to the Sipbroker PSTN number whereas all the 2 way ones have a different greeting (female)
So is there some difference in the boxes being used at the Sipbroker hosts?

What I am trying to acheive is a shared DID facility reacheable from an Australian number

martin 01-13-2010 09:58 PM

Some of the DIDs re invite the SIP session to optimize audio. Based on your symptoms I suspect this renegotiation is causing problems for your ATA.

I am not familiar with your device. However, I would be looking for a setting like "SIP REINVITE" and disabling it.

rogerhilary 01-16-2010 05:22 PM

Martin

Thank you for that speedy response.
I have trawled through the manula for the Linksys PAP2 (as well as its bigger brother the Sipura SPA 3000) and can find nothing to enable or disable facilities for SIP re-invites.

I then looked through this forum and it seems that others have had issues with SIP re-invites and they too have not been able to "switch them off". I see that the server which initiates the SIP re-invite is supposed to first enquire if the UA does support SIP re-invites and only if it does issue one.
On this basis something seems to be broken, either the server is not asking, the PAP2 is responding incorrectly or the server is not taking "cannot handle rre-invites" into account. I have not been able in the syslog of the SIP messages to recognise the server asking about SIP re-invites, it just seems to do it, but then again maybe I missed it.


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