Posted: Thu May 21, 2009 8:06 am Post subject: [asterisk-users] Jitter buffer question
Hi List,
I have a question regarding jitterbuffer in Asterisk 1.4.24. I see that
jitterbuffer is only effective on the receiving channels.
My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch
office.
Questions:
1. To enable jitter buffer on SIP channels it seems I have to enable and
force it, right?
2. If I enable and force jitter buffer, Asterisk would always have to
stay in media path to make it function, right? If I am right, this
effectively disables native RTP bridging.
3. Is it possible to only enable jitter buffer on calls where the SIP
trunk is involved? It is no use for me to enable the jitter buffer
between SIP phones on the same LAN.
Many thanks for all answers, I have tried hard to google out them, but
no success so far.
Ondrej
_______________________________________________
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I have a question regarding jitterbuffer in Asterisk 1.4.24. I see
that
jitterbuffer is only effective on the receiving channels.
My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch
office.
Questions:
1. To enable jitter buffer on SIP channels it seems I have to enable
and
force it, right?
Not sure about the forcing part (don't know exacly how it works), but I always set jbforce=yes to be sure.
Quote:
2. If I enable and force jitter buffer, Asterisk would always have to
stay in media path to make it function, right? If I am right, this
effectively disables native RTP bridging.
Yes, there's no way Asterisk can create buffers if it's not on the media path.
Quote:
3. Is it possible to only enable jitter buffer on calls where the SIP
trunk is involved? It is no use for me to enable the jitter buffer
between SIP phones on the same LAN.
Sure, just put the jbenable and other options on the SIP section of that trunk, instead of putting it on [general].
Quote:
Many thanks for all answers, I have tried hard to google out them, but
no success so far.
Ondrej
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Posted: Thu May 21, 2009 12:30 pm Post subject: [asterisk-users] Jitter buffer question
Hi Vinicius.
Quote:
>/ 1. To enable jitter buffer on SIP channels it seems I have to enable
/>>/ and
/>>/ force it, right?
/
Quote:
Not sure about the forcing part (don't know exacly how it works), but I always set jbforce=yes to be sure.
Ok, thanks!
Quote:
>/ 2. If I enable and force jitter buffer, Asterisk would always have to
/>>/
/>>/ stay in media path to make it function, right? If I am right, this
/>>/ effectively disables native RTP bridging.
/
Quote:
Yes, there's no way Asterisk can create buffers if it's not on the media path.
Yes, that makes a sense. I was just wondering if it is possible to configure it the
way that the jitterbuffer is enabled only if the asterisk server can not do native RTP bridging...
Quote:
>/ 3. Is it possible to only enable jitter buffer on calls where the SIP
/>>/
/>>/ trunk is involved? It is no use for me to enable the jitter buffer
/>>/ between SIP phones on the same LAN.
/
Quote:
Sure, just put the jbenable and other options on the SIP section of that trunk, instead of putting it on [general].
Well, I think that would not work since the jitterbuffer is only effective on the outgoing channels.
If I receive a call from the SIP trunk, I hear jitter. To suppress it, I would have to enable jbforce/jbenable on my
local SIP channel as this is the outgoing one - the SIP trunk is the incoming one, right?
Many thanks,
Ondrej
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>/ 1. To enable jitter buffer on SIP channels it seems I have to
enable
/>>/ and
/>>/ force it, right?
/
> Not sure about the forcing part (don't know exacly how it works),
but I always set jbforce=yes to be sure.
Ok, thanks!
>>/ 2. If I enable and force jitter buffer, Asterisk would always have
to
/>>/
/>>/ stay in media path to make it function, right? If I am right,
this
/>>/ effectively disables native RTP bridging.
/
> Yes, there's no way Asterisk can create buffers if it's not on the
media path.
Yes, that makes a sense. I was just wondering if it is possible to
configure it the
way that the jitterbuffer is enabled only if the asterisk server can
not do native RTP bridging...
>>/ 3. Is it possible to only enable jitter buffer on calls where the
SIP
/>>/
/>>/ trunk is involved? It is no use for me to enable the jitter
buffer
/>>/ between SIP phones on the same LAN.
/
> Sure, just put the jbenable and other options on the SIP section of
that trunk, instead of putting it on [general].
Well, I think that would not work since the jitterbuffer is only
effective on the outgoing channels.
If I receive a call from the SIP trunk, I hear jitter. To suppress it,
I would have to enable jbforce/jbenable on my
local SIP channel as this is the outgoing one - the SIP trunk is the
incoming one, right?
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