SipXtapi library voice quality issue

[Sathyanarayanan Ramakrishnan]

Hi,

We are facing issue in SipXtapi library when taking calls. Calls are randomly degrading with static voice and under water effect and it is very random. Out of 50 calls, 2 to 3 calls would be underwater and quality would be so poor to understand both the ends. We are having following configurations in the code for every call:

1. Audio Codec : G711
2. Audio Bandwidth: High
3. Noise Reduction mode: High
4. AEC mode : Cancel


It would be appreciated for any suggestion on debugging the voice quality issue that we face in our environment.

Thanks,
Sathya

[Daniel Petrie]

Hi Sathya:

Just to be sure that we are talking about the same branch and not some other fork, are you using:

http://scm.sipfoundry.org/rep/sipX/main

 

The vanilla, open source version of sipX does not have an adaptive jitter buffer or a very robust packet loss compensation algorithm.  It you have network situations where you have any sort of packet loss or jitter, you probably will want to consider using SIPez's proprietary media engine plugin (SME) for sipX.  With SIPez's SME, packet loss is not noticible to most people with up to 5% packet loss which is a pretty huge amount of loss for most network situations.  SIPez's SME also minimizes latency which can build up sometimes with just the pure open source sipX solution.  SIPez's SME is a pay for/commercial plugin.

Cheers,

Dan

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[Sathyanarayanan Ramakrishnan]

We are using sipXtapi 3.2.0 version and it is not the same as the branch specified below I believe. I shall try the given below main branch and see whether it improves.

Is this the bug fix version of SipXtapi 3.2.0?