VoIP Mailing List Archives

[kokoska.rokoska at pos...]

Hi all,

I fall into trouble with voice mail. It looks like FreeSWITCH sends no
RTP during Voice Mail recording and thus the calls from my TSPs are cut
off in the middle of the recording due to lack of RTP activity (based on
providers "tolerancy" it happens in 5 to 20 seconds).

I tried to set VAD to "none" in all sofia profiles but it doesn't help.
Are there any other settings I have to use to force FreeSWITCH to send
RTP back (silence, CNG or what ever during VM recording?

BTW: I'm on current trunk.

Any hint is very welcome

Best regards,

kokoska.rokoska

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[kokoska.rokoska at pos...]

kokoska.rokoska napsal(a):

[kokoska.rokoska at pos...]

Anthony Minessale napsal(a):

[anthony.minessale at g...]

it's nothing to do with vad, it's simply how FS works.

It's a waste to encode and send zeros into the channel while it's recording.
Also, It's unreasonable to have such a short timeout.

I understand it's not your fault, I am just letting you know.

It would be possible to add a patch to create a channel variable like
NDLB_waste_bandwidth_while_recording or something but it does not exist today.


On Mon, Apr 20, 2009 at 11:03 AM, kokoska rokoska <kokoska.rokoska@post.cz (kokoska.rokoska@post.cz)> wrote:

[chris.chen2004 at gmai...]

Hi kokoska
Actually, you can request your VSP to set the rtptimeout or whatever parameter in their SIP server to a reasonable value such as 300 seconds as 5 minutes should be enough for most standard business voice mail service, otherwise you should wait for live calls instead of leaving voice messages.

In * they have the following setting which is default to 60 seconds if nothing changed

rtptimeout=300                  ; Terminate call if 60 seconds of no RTP or RTCP activity
                                ; on the audio channel
                                ; when we're not on hold. This is to be able to hangup
                                ; a call in the case of a phone disappearing from the net,
                                ; like a powerloss or grandma tripping over a cable.

This works with one of my ITSP as they provide SIP trunking via *

Hope this helps.

Chris


On Mon, Apr 20, 2009 at 12:48 PM, kokoska rokoska <kokoska.rokoska@post.cz (kokoska.rokoska@post.cz)> wrote:

[kokoska.rokoska at pos...]

Thank you very much, Chris, for your reply!


Chris Chen napsal(a):

[kokoska.rokoska at pos...]

Anthony Minessale napsal(a):

[dujinfang at gmail.com]

You are not alone, I vote 1.

And there's a similer variable in conference:

<!--Can be | delim of waste|mute|deaf waste will always
transmit data to each channel
even during silence -->
<!--<param name="member-flags" value="waste"/>-->

On Apr 24, 2009, at 1:40 PM, kokoska rokoska wrote:

[kokoska.rokoska at pos...]

seven napsal(a):

[anthony.minessale at g...]

sigh,

see r13144


On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska <kokoska.rokoska@post.cz (kokoska.rokoska@post.cz)> wrote:

[kokoska.rokoska at pos...]

Thank you very much, Anthony, for such fast solution!

May I ask you - How should I activate this feature?
I have tried to "grep" through sources for new NDLB variable but I
didn't find one...

Best regards,

kokoska.rokoska

Anthony Minessale napsal(a):

[dujinfang at gmail.com]

I haven't tested but I guess it's just like other variables and I
documented to here:

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


On Apr 25, 2009, at 11:56 PM, kokoska rokoska wrote:

[kokoska.rokoska at pos...]

Thank you very much, dujinfang, for your help!

When I use
<action application="set" data="record_waste_resources=true"/>

the FreeSWITCH really sends back RTP stream during recording, but
instead of (faked) silence it is full of completely regular load noise
I have tested it with different devices (Linskys, Snom, FritzBOX,
Nokia...) with the same result (even pcap files looks similar).

Dialplan snipped looks like:
<action application='answer'/>
<action application='playback' data='silence_stream://1000'/>
<action application='set' data='record_waste_resources=true'/>
<action application='voicemail' data='context $${domain} number'/>
<action application='hangup'/>

Do you (or anybody else know what I'm doing wrong?

Thanks once more, dujinfang, for your help!

Best regards,

kokoska.rokoska



dujinfang napsal(a):

[dave at 3c.co.uk]

Add something like
memset(write_buf, 0, SWITCH_RECOMMENDED_BUFFER_SIZE);
after
char write_buf[SWITCH_RECOMMENDED_BUFFER_SIZE];
in switch_ivr_play_say.c (line 395)

--Dave

[kokoska.rokoska at pos...]

Thank you very much, Dave, for your help!

Mentioned modification did the trick and all works as I wish Thank you!

BTW: When I look into the pcap files for RTP stream, I see some strange
timing for outgoing RTP from FreeSWITCH. Every even packet is sent after
24 ms and every odd packet at 12 ms...
"ptime" is on both sides set to 20 ms, incomming stream has nearly no
jitter, server is on real HW (no virtualization) and almost idle (the
most consuming process is "htop"
Is it normal or should I investigate what is wrong?

Best regards,

kokoska.rokoska


David Knell napsal(a):