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Not responding to RTCP PINGs #51

fallenpegasus opened this Issue Mar 22, 2012 · 13 comments


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4 participants

When on long calls into my corporate provided SIP PBX, the call hangs up after half an hour. According to the administrator of our SIP servers, they send a RTCP PING every 5 minutes, and after not getting ping response after 6 tries, their gateway shuts the connection down.

So, please make Telephone.app respond to RTCP PINGs.


Hello, there have been several releases of Telephone since this issue was created. Could you please download and run the latest version to confirm this issue still exists.

Sorry for the template response, just trying to clean up and close old issues here.


Yes, issue still there in last version 1.1.4.

SIP Session-Timers (RFC 4028) provide an end-to-end keep-alive mechanism for active SIP sessions.
This mechanism can detect and reclaim SIP channels that do not terminate through normal signaling procedures.

PJSIP supports Session-Timers, however perhaps they are not yet implemented in Telephone ?

In Asterisk, the related parameter (in sip.conf) is session-expires (in seconds, 1800 as the default value).
As a (not nice at all) workaround, it can be increased to a higher value...

Thank you !

Alexei, do you think you could handle SIP Session-Timers in the future release please ?
That's sometimes embarrassing to have the call cut during the conversation :|
Thank you very much !


eofster commented Nov 17, 2015

The next bug fixing release, 1.1.5, should include the latest stable PJSIP release. I haven’t checked it for the feature in question yet, but if it is there and enabled by default, it should just work.

I'm running one of your last commits with PJSIP 2.4.5 since a few days.
Unfortunately issue is still present, call hangs up after the Asterisk session-expires parameter.
Thank you Alexei 👍


eofster commented Nov 17, 2015

Okay, thank you for the info.

@eofster eofster added this to the 1.1.5 milestone Nov 17, 2015

@eofster eofster added the bug label Nov 17, 2015

@eofster eofster self-assigned this Nov 17, 2015


eofster commented Mar 23, 2016

@fallenpegasus Mark, if you’re still here and interested, could you please ask the administrator to provide some more information about RTCP PINGs? What exactly is sent and what exactly is expected to be received? I couldn’t find the term “RTCP PING” in the RFC 3550. Could it be some custom configuration that is not described in the RFCs?

@eofster, sounds like they are documented in RFC 4028 (SIP Session-Timers).
Sounds like PJSIP handles Session-Timers, perhaps you need to "activate" them ?


eofster commented Mar 23, 2016

@benrubson Ben, let’s discuss the SIP session timers problem in #180.

Don't you think these 2 issues are the same ?


eofster commented Mar 23, 2016

They might be, but let’s separate them to avoid overlapped discussions.


eofster commented Mar 23, 2016

Sounds like PJSIP handles Session-Timers, perhaps you need to "activate" them ?

That’s exactly the strange part. SIP session timers seem to be activated in PJSIP by default. It itself sets the header when making calls.


eofster commented Mar 29, 2016

In the originally reported problem calls hang up after 30 minutes. This seems very much like the default limit of SIP session timers, and not like a timer related to RTP, which usually depends on voice being transmitted. I suspect that the original problem could be incorrectly diagnosed and it might be fixed in #180.

Closing this issue for now, but if my suspicion is incorrect and the behavior stays the same after Telephone 1.1.5 is released, please reopen the issue.

@eofster eofster closed this Mar 29, 2016

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