End call action lag
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Originally created by: a.emerso...@gmail.com
Originally owned by: r3gis...@gmail.com
What steps will reproduce the problem?
1.Place SIP call through PBXES.org
2.End call
3.
What is the expected output? What do you see instead?
End call action is delayed by 5-10 seconds i.e. the call continues after the end call action was initiated
What version of the product are you using? On what operating system?
Version 0.00-13 on HTC Nexus One/Cyanogenmod 6(Custom Froyo rom)
Please provide any additional information below.
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Originally posted by: r3gis...@gmail.com
Is it reproducible over Wifi or 3G only (or GPRS/EDGE)?
Actually, CSipSimple doesn't show "End call" while caller/e send a ACK for BYE. If ACK take a while to be received, it can explain 5-10 seconds of delay.
Over 3G (and more over GPRS) data packet are treated turn by turn.
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Originally posted by: a.emerso...@gmail.com
This was tested on Wifi
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Originally posted by: r3gis...@gmail.com
So, that's strange.
You reproduce for each call and to several contacts?
Or it's intermittent / only one callee (/or sip softphone target).
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Originally posted by: a.emerso...@gmail.com
I can reproduce this for each call to any contact, provided that the call gets connected.
Before the call gets connected, I can end the call with no delay
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Originally posted by: mcampbel...@gmail.com
I have exactly the same issue on my X10 with android 1.6 and HTC Desire with OpenDesire.
I am not using pbxes but my own pbx, so its not a pbxes only issue.
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Originally posted by: a.emerso...@gmail.com
Hi mcampbell,
Have you been able to reproduce this recently? I wasn't able to with calls I made in the last few hours.
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Originally posted by: lubomir....@gmail.com
I also can't reproduce this issue with PlanetPhone.net services, cSIPsimple latest dev version here with OpenDesire ROM on HTC Desire
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Originally posted by: mcampbel...@gmail.com
I still can reproduce this. Latest dev version, OpenDesire ROM on HTC Desire.
What logs can I provide?
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Originally posted by: r3gis...@gmail.com
(No comment was entered for this change.)
Mergedinto: 200
Status: Duplicate
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Originally posted by: r3gis...@gmail.com
(No comment was entered for this change.)
Mergedinto: -200
Status: Accepted
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Originally posted by: pathak.a...@gmail.com
I am unable to end call and sometime suddenly call get disconnected while talking.
Sometimes even fails to register with provider.
I am using Nexus one and voip provider is localphone.com
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Originally posted by: pgrayb...@gmail.com
Same here. 5-10 second delay from pressing red end call button to disconnect. EVO 2.2, aptela.com
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Originally posted by: adamsim...@gmail.com
I can confirm the same issue on a HTC Evo with 2.2 and using pbxes.org connected to a SipGate account.
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Originally posted by: hussein....@gmail.com
Confirm Issue on samsung galaxy i9000 using own pbx or pbxes.org, however it is intermittent.
Are there any logs useful for you?
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Originally posted by: mccor...@gmail.com
My inability to hang up is often accompanied by messages like this:
E/libpjsip( 1320): 12:06:59.070 sip_endpoint.c Error processing packet from 203.25.26.98:5060: Unknown error 171050 Via [code 171050]:
E/libpjsip( 1320): SIP/2.0 487 Request Terminated
E/libpjsip( 1320): Record-Route: <sip:203.25.26.98;lr;did=7dd.da60db57>
E/libpjsip( 1320): To: <sip:0403399386@sip.vtel.biz>;tag=39c9f168-co10414-INS001
E/libpjsip( 1320): From: <sip:999000431@sip.vtel.biz>;tag=rQcZAjlHaKfpfqT4skxWyfKPUXecgyLD
E/libpjsip( 1320): Call-ID: 2Wo8SRztP6HVfKTobQel7FhkZuKKWPnV
E/libpjsip( 1320): CSeq: 13947 INVITE
E/libpjsip( 1320): User-Agent: ENSR3.0.63.1-IS1-RMRG371-RG8501-CPO6970
E/libpjsip( 1320): Content-Length: 0
E/libpjsip( 1320):
E/libpjsip( 1320):
E/libpjsip( 1320): -- end of packet.
E/libpjsip( 1320): 12:07:01.132 pjsua_call.c Failed to send end session message: Object already exists (PJ_EEXISTS) [status=70015]
Phone = Nexus One.
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Originally posted by: r3gis...@gmail.com
Ok, this log is probably about a second try to end the call.
My understand of this issue is that for some unknown reason, after the ACK of the call, pjsip tries to contact directly the remote contact IP without trying to use its sip proxy.
And so as a result everything that is tried to be sent to the remote contact (including the BYE request) is send to a private IP... which, of course fail.
I've to find out why pjsip behaves as it
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Originally posted by: dan.harl...@gmail.com
I often experience an extremely long hangup delay on my Samsung Transform, sometimes on the order of a minute or longer. Happens whether connected via Wifi or 3G. When the far-end disconnects, it responds immediately.
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Originally posted by: r3gis...@gmail.com
Issue 358 has been merged into this issue.
Related
Tickets:
#358View and moderate all "tickets Discussion" comments posted by this user
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Originally posted by: r3gis...@gmail.com
Good news... I know what it is doing that with some sip provider...
I'm fixing it, and it will be available in next build. :)
For reference, was something about sip uri : I was dialing sip:username@domain.loc, while <sip:username@domain.loc> seems to be more supported by these providers!
Owner: r3gis.3R
Status: Started
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Originally posted by: r3gis...@gmail.com
Should be fixed in 0.00-15-11 available on download section.
Status: Fixed