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
Bruno Rocha wrote:> Hello everybody,Hola,> 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
Possibly Parallel Threads
- Multi-line phones - Asterisk uses wrong callerid
- OT - Aastra phones provisioning
- Aastra phones and park/pickup feature
- Aastra phones
- Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?