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bruno at 3gnt.net
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 Posted: Mon Jul 14, 2014 11:12 am Post subject: [asterisk-users] Call drop on Aastra SIP phones |
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Hello everybody,
I'm having issues with calls being dropped on Aastra phones, when the
call is on hold. Tested with models 6863i and 6867i.
I've figured that the call is dropped by Asterisk when it reaches the
rtpholdtimeout limit.
I've reported the issue to Aastra, asking them to implement some kind of
"RTP keep-alive" feature on their phones. Maybe the phone could send
some RTCP frame (or an empty RTP frame) just to prove it is alive.
Unfortunately Aastra said the hold behaviour on the phone is correct, as
per RFC 3264, section 8.4, 4th paragraph:
Typically, when a user "presses" hold, the agent will generate an
offer with all streams in the SDP indicating a direction of sendonly,
and it will also locally mute, so that no media is sent to the far
end, and no media is played out.
They can implement the "RTP keep-alive" feature only if there is some
RFC describing that behaviour.
Is Aastra correct? Should I configure Asterisk with rtpholdtimeout=0 to
solve this issue and make Asterisk RFC compliant?
Or should Asterisk rtpholdtimeout code take the phone RFC behaviour into
account and don't drop the call?
Best regards,
--
Bruno Rocha
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jcolp at digium.com
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| Posted: Tue Jul 15, 2014 7:39 am Post subject: [asterisk-users] Call drop on Aastra SIP phones |
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Bruno Rocha wrote:
Hola,
| Quote: | I'm having issues with calls being dropped on Aastra phones, when the
call is on hold. Tested with models 6863i and 6867i.
I've figured that the call is dropped by Asterisk when it reaches the
rtpholdtimeout limit.
I've reported the issue to Aastra, asking them to implement some kind of
"RTP keep-alive" feature on their phones. Maybe the phone could send
some RTCP frame (or an empty RTP frame) just to prove it is alive.
Unfortunately Aastra said the hold behaviour on the phone is correct, as
per RFC 3264, section 8.4, 4th paragraph:
Typically, when a user "presses" hold, the agent will generate an
offer with all streams in the SDP indicating a direction of sendonly,
and it will also locally mute, so that no media is sent to the far
end, and no media is played out.
|
They are correct. The "rtpholdtimeout" option stems from a time when it
was not possible to monitor the signaling of the call and is an
Asterisk-ism. You've got a few options, though:
1. Increase the rtpholdtimeout
2. Don't use rtpholdtimeout and use SIP session timers instead (check
the SIP Session-Timers section in sip.conf.sample)
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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bruno at 3gnt.net
Guest
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| Posted: Tue Jul 15, 2014 9:21 am Post subject: [asterisk-users] Call drop on Aastra SIP phones |
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Hi Joshua!
On 2014-07-15 13:39, Joshua Colp wrote:
| Quote: | Bruno Rocha wrote:
Hola,
| Quote: | I'm having issues with calls being dropped on Aastra phones, when the
call is on hold. Tested with models 6863i and 6867i.
I've figured that the call is dropped by Asterisk when it reaches the
rtpholdtimeout limit.
I've reported the issue to Aastra, asking them to implement some kind of
"RTP keep-alive" feature on their phones. Maybe the phone could send
some RTCP frame (or an empty RTP frame) just to prove it is alive.
Unfortunately Aastra said the hold behaviour on the phone is correct, as
per RFC 3264, section 8.4, 4th paragraph:
Typically, when a user "presses" hold, the agent will generate an
offer with all streams in the SDP indicating a direction of sendonly,
and it will also locally mute, so that no media is sent to the far
end, and no media is played out.
|
They are correct. The "rtpholdtimeout" option stems from a time when it
was not possible to monitor the signaling of the call and is an
Asterisk-ism. You've got a few options, though:
1. Increase the rtpholdtimeout
2. Don't use rtpholdtimeout and use SIP session timers instead (check
the SIP Session-Timers section in sip.conf.sample)
|
Thanks for the clarification! I will try the SIP Session-Timers.
Cheers,
--
Bruno Rocha
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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