[Rtcomm] [Bug 7302] New: SIP over wifi unusable on N900 due to choppy audio

[bugzilla-daemon at maemo.org]

https://bugs.maemo.org/show_bug.cgi?id=7302

           Summary: SIP over wifi unusable on N900 due to choppy audio
           Product: Chat & Call & SMS
           Version: 5.0/(1.2009.42-11)
          Platform: N900
        OS/Version: Maemo
            Status: UNCONFIRMED
          Severity: normal
          Priority: Medium
         Component: SIP
        AssignedTo: rtcomm at maemo.org
        ReportedBy: pierre-marc.fournier at polymtl.ca
         QAContact: sip-bugs at maemo.bugs


SOFTWARE VERSION:
Maemo 5 (1.2009.42-11.002)

EXACT STEPS LEADING TO PROBLEM: 
(Explain in detail what you do (e.g. tap on OK) and what you see (e.g. message
Connection Failed appears))
1. Connect to network using wifi.
2. Make an outgoing SIP call.

EXPECTED OUTCOME:
Audio quality should be good in both directions.

ACTUAL OUTCOME:
The call is established correctly. The N900 user can hear the person he is
calling fine. However, the person being called hears very bad audio quality:
the voice of the caller varies from very choppy to inaudible.

REPRODUCIBILITY:
Always, in a number of different setups (see below).

EXTRA SOFTWARE INSTALLED:
None, this was observed on a pristine N900 owned for less than 10 minutes.

OTHER COMMENTS:

- Tried different SIP servers (one Asterisk, one BroadWorks) with different
providers.
- Tried rebooting the N900
- Tried different wifi networks on different internet connections (both are
broadband and have other sip phones working well on them on these same
providers)
- Tried different wifi security types (WPA2, WEP, none)
- Tried disabling wifi power saving
- Tried high and low transmission power settings

All these setups resulted in the same problem.

The traffic was captured on the system running Asterisk. The audio coming from
the N900 and going in to asterisk is fine, but the audio coming out of asterisk
is bad. Therefore I presume asterisk must be doing something to the audio.
However, BroadWorks, a very different PBX, is doing the same thing, so it
appears to be the N900's fault. Could it be some RTP timestamping issue? (I am
not a voip expert.)

I also noticed that for each RTP frame going in to asterisk from the N900,
asterisk generates two frames of half the length for sending to the recipient.
A bit of research on posts concerning other devices tells that this can yield
bad audio quality in some cases. Could this be the problem?

Audio quality is good when making a call using gtalk or skype.



User-Agent:       Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.9.1.5)
Gecko/20091112 Iceweasel/3.5.5 (like Firefox/3.5.5; Debian-3.5.5-1)


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