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[peter.olsson at vision...]

After using the latest trunk revisions I get no audio anymore. The last working build I have is about 5 days ago. I havn’t upgraded until today, so I don’t know exactly when this happened.

I’ve noticed quite a few changes on the RTP stack, beacuse of the implementation om ZRTP, and I guess it’s somewhere around this time when it happened. How to continue debugging on this issue? I have both a working version of FS (compiled 5 days ago), and a broken one (compiled today), so I can test this very easily, and everything is on a non live server.

The conf-dir is the same between the revisions.

The calls I’m trying to do is both directly to FS (voicemail or similar applications), and aslo calls to another SIP-trunk, to PSTN (media stream is sent through FS).

Regards,

Peter

[anthony.minessale at g...]

the upgrade changed the switch_frame structure so most likely you did not do "make current" as we always recommend.
Try that......


On Fri, May 29, 2009 at 5:46 AM, Jason White <jason@jasonjgw.net (jason@jasonjgw.net)> wrote:

[brian at freeswitch.org]

Nope its not a sofia issue... its build skew

/b

On May 29, 2009, at 8:24 AM, Peter Olsson wrote:

[brian at freeswitch.org]

Its called build skew... we added an extra_data element to the frame struct. Please do a fresh checkout and build.

/b

On May 29, 2009, at 5:46 AM, Jason White wrote:

[anthony.minessale at g...]

when you say "i did that"
you typed "make current" to rebuild?

or you are assuming your successful compile is the same effect as cleaning the 100 object files
that have the wrong data structure in them so the audio data they really seek is 8 bytes offset from where they think they are
until they are deleted and recompiled by the make current command?

The coincidental side-effect of this is no audio in any rtp streams...... *shrug* or you can continue to update more and more revisions
on top of each other and end up with even worse build skew.

We tried 3 times to tell you that's plenty......
 


On Fri, May 29, 2009 at 8:28 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:

[anthony.minessale at g...]

did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ?


On Fri, May 29, 2009 at 8:33 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:

[peter.olsson at vision...]

I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything.

/Peter

-----Ursprungligt meddelande-----
Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Jason White
Skickat: den 29 maj 2009 12:47
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Peter Olsson <peter.olsson@visionutveckling.se> wrote:

[mike at jerris.com]

Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org.

Thanks
Mike

On May 29, 2009, at 9:55 AM, Peter Olsson wrote:

[peter.olsson at vision...]

I'll try to do this during this weekend.

I've looked through the SNV logs, and I really can't find a good reason for this to happen. And when looking into wireshark I can see RTP audio flowing from FS to my SIP phone, but not in the other direction. So this still makes me wonder if something has happened to sofia (that sets up the media incorrectly)... And also when I hangup the call, it takes about a minute for FS to detect this, and it reports hangup reason unknown.

But as I said, I'll look into this a bit deeper during this weekend, and file a jira case when I have some more information.


//Peter

________________________________________
Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Michael Jerris [mike@jerris.com]
Skickat: den 29 maj 2009 19:35
Till: freeswitch-users@lists.freeswitch.org
Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org.

Thanks
Mike

On May 29, 2009, at 9:55 AM, Peter Olsson wrote:

I’m on Windows, so I have everything under my fs directory, but I deleted the complete directory and did everything from scratch...

/Peter


Från: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale
Skickat: den 29 maj 2009 15:46
Till: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ?

On Fri, May 29, 2009 at 8:33 AM, Peter Olsson <peter.olsson@visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote:

Nope – it’s not

Just to make sure I even deleted the source completely, and checked everything out again.

Från: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Brian West
Skickat: den 29 maj 2009 15:26

Till: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Nope its not a sofia issue... its build skew

On May 29, 2009, at 8:24 AM, Peter Olsson wrote:

I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything.

/Peter


!DSPAM:4a201fa932931035648682!

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[peter.olsson at vision...]

Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end.

/Peter


-----Ursprungligt meddelande-----
Från: Peter Olsson
Skickat: den 30 maj 2009 09:01
Till: freeswitch-users@lists.freeswitch.org
Ämne: FW: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

And just to be clear, even though media flows in one direction (from FS to phone), I get no audio at all.

And by the way, I mean SVN, not SNV Sorry for double posting...

/Peter

________________________________________
Från: Peter Olsson
Skickat: den 30 maj 2009 08:58
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

I'll try to do this during this weekend.

I've looked through the SNV logs, and I really can't find a good reason for this to happen. And when looking into wireshark I can see RTP audio flowing from FS to my SIP phone, but not in the other direction. So this still makes me wonder if something has happened to sofia (that sets up the media incorrectly)... And also when I hangup the call, it takes about a minute for FS to detect this, and it reports hangup reason unknown.

But as I said, I'll look into this a bit deeper during this weekend, and file a jira case when I have some more information.


//Peter

________________________________________
Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Michael Jerris [mike@jerris.com]
Skickat: den 29 maj 2009 19:35
Till: freeswitch-users@lists.freeswitch.org
Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org.

Thanks
Mike

On May 29, 2009, at 9:55 AM, Peter Olsson wrote:

I'm on Windows, so I have everything under my fs directory, but I deleted the complete directory and did everything from scratch...

/Peter


Från: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale
Skickat: den 29 maj 2009 15:46
Till: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ?

On Fri, May 29, 2009 at 8:33 AM, Peter Olsson <peter.olsson@visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote:

Nope - it's not

Just to make sure I even deleted the source completely, and checked everything out again.

Från: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Brian West
Skickat: den 29 maj 2009 15:26

Till: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>
Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Nope its not a sofia issue... its build skew

On May 29, 2009, at 8:24 AM, Peter Olsson wrote:

I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything.

/Peter


!DSPAM:4a201fa932931035648682!

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[kristian.kielhofner at...]

Peter,

Ouch. Your PBX is broken. It shouldn't do that.

Luckily FreeSWITCH provides a way to select RPID/PAI/none:

http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation

On Mon, Jun 1, 2009 at 5:57 AM, Peter Olsson
<peter.olsson@visionutveckling.se> wrote:

[brian at freeswitch.org]

No he's talking about the one from SFSIP-111, Again that shouldn't
matter... I'll fix that.

/b

On Jun 1, 2009, at 8:19 AM, Kristian Kielhofner wrote:

[peter.olsson at vision...]

Thanks for the reply!

Will this really help though? From what I understand of the change that breaks the compatibility, it will always send this header in the "200 OK" message (my problem is incoming calls to FS). The fix you're describing, isn't it when calling from FS to the other end? This way it works either way, it's just when the PBX gets this in the 200 OK message with this header that it stops working.

/Peter

-----Ursprungligt meddelande-----
Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Kristian Kielhofner
Skickat: den 1 juni 2009 15:20
Till: freeswitch-users@lists.freeswitch.org
Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

Peter,

Ouch. Your PBX is broken. It shouldn't do that.

Luckily FreeSWITCH provides a way to select RPID/PAI/none:

http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation

On Mon, Jun 1, 2009 at 5:57 AM, Peter Olsson
<peter.olsson@visionutveckling.se> wrote:

[brian at freeswitch.org]

I'll fix it where it won't do it on anything but polycom.

/b

On Jun 1, 2009, at 8:32 AM, Peter Olsson wrote:

[brian at freeswitch.org]

Your PBX is broken but now I have fixed it to only do that feature with Polycom for now.

/b

On Jun 1, 2009, at 8:32 AM, Peter Olsson wrote: