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cursor at telecomab.mx
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 Posted: Mon Sep 07, 2020 7:36 pm Post subject: [asterisk-users] Some calls drop after 30 seconds |
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Some users have complained that their calls drop after about 30
seconds. Not all, just some. After looking at the log files the only
difference I can find from the dropped calls is the following line:
[2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge
14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge
technology to native_rtp
Most calls just do:
[2020-09-07 18:13:56] VERBOSE[15293][C-00000084] bridge_channel.c:
Channel PJSIP/1028-0000012a joined 'simple_bridge' basic-bridge
<626258fc-0649-45c7-b0d3-630a06d2c91b>
Why are some calls using the simple bridge and others switch to the
native_rtp bridge? Could this be a codec problem? How can I prevent
the switch?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161
--
_____________________________________________________________________
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Check out the new Asterisk community forum at: https://community.asterisk.org/
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jcolp at sangoma.com
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| Posted: Tue Sep 08, 2020 4:17 am Post subject: [asterisk-users] Some calls drop after 30 seconds |
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On Mon, Sep 7, 2020 at 9:35 PM Carlos Chavez <cursor@telecomab.mx (cursor@telecomab.mx)> wrote:
| Quote: | Some users have complained that their calls drop after about 30
seconds. Not all, just some. After looking at the log files the only
difference I can find from the dropped calls is the following line:
[2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge
14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge
technology to native_rtp
Most calls just do:
[2020-09-07 18:13:56] VERBOSE[15293][C-00000084] bridge_channel.c:
Channel PJSIP/1028-0000012a joined 'simple_bridge' basic-bridge
<626258fc-0649-45c7-b0d3-630a06d2c91b>
Why are some calls using the simple bridge and others switch to the
native_rtp bridge? Could this be a codec problem? How can I prevent
the switch?
|
It depends on the channels involved as well as the features in use. To prevent direct media from occurring you can set the "direct_media" option to "no" on the endpoint. The native_rtp bridge can still be used, though, to provide more efficient in-Asterisk forwarding of media.
If that doesn't change things you'd need to examine further, such as looking at the SIP trace for a call (pjsip set logger on) as 30 seconds is close to the amount of time for a lost ACK to a 200 OK, which generally indicates a NAT issue.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org |
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duncan at turnbull.co.nz
Guest
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| Posted: Tue Sep 08, 2020 5:59 am Post subject: [asterisk-users] Some calls drop after 30 seconds |
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Hi Carlos
On Tue, 8 Sep 2020, 12:36 pm Carlos Chavez, <cursor@telecomab.mx (cursor@telecomab.mx)> wrote:
| Quote: | Some users have complained that their calls drop after about 30
seconds. |
The rtp timeout is usually about 30 seconds. If rtp is only 1 way then the calls will drop after 30 secs. This is usually nat/firewall related so a packet dump helps to confirm. I also find using tcpdump to write a pcap file that I can feed into wireshark is helpful as wireshark has great sip decoding options. It will trace the callflow, pull out relevant packets, replay audio. Its very helpful
Is there anything different about these users and their setup? Or who they are calling?
| Quote: | Not all, just some. After looking at the log files the only
difference I can find from the dropped calls is the following line:
[2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge
14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge
technology to native_rtp
Most calls just do:
[2020-09-07 18:13:56] VERBOSE[15293][C-00000084] bridge_channel.c:
Channel PJSIP/1028-0000012a joined 'simple_bridge' basic-bridge
<626258fc-0649-45c7-b0d3-630a06d2c91b>
Why are some calls using the simple bridge and others switch to the
native_rtp bridge? Could this be a codec problem? How can I prevent
the switch?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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cursor at telecomab.mx
Guest
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| Posted: Tue Sep 08, 2020 10:46 am Post subject: [asterisk-users] Some calls drop after 30 seconds |
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On 08/09/20 4:16, Joshua C. Colp wrote:
| Quote: | On Mon, Sep 7, 2020 at 9:35 PM Carlos Chavez <cursor@telecomab.mx (cursor@telecomab.mx)> wrote:
| Quote: | Some users have complained that their calls drop after about 30
seconds. Not all, just some. After looking at the log files the only
difference I can find from the dropped calls is the following line:
[2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge
14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge
technology to native_rtp
Most calls just do:
[2020-09-07 18:13:56] VERBOSE[15293][C-00000084] bridge_channel.c:
Channel PJSIP/1028-0000012a joined 'simple_bridge' basic-bridge
<626258fc-0649-45c7-b0d3-630a06d2c91b>
Why are some calls using the simple bridge and others switch to the
native_rtp bridge? Could this be a codec problem? How can I prevent
the switch?
|
It depends on the channels involved as well as the features in use. To prevent direct media from occurring you can set the "direct_media" option to "no" on the endpoint. The native_rtp bridge can still be used, though, to provide more efficient in-Asterisk forwarding of media.
If that doesn't change things you'd need to examine further, such as looking at the SIP trace for a call (pjsip set logger on) as 30 seconds is close to the amount of time for a lost ACK to a 200 OK, which generally indicates a NAT issue.
|
Direct media is off for all endpoints (both trunks and phones). There is no NAT on either side, the phones are on the local network and the trunk provider has a direct link and the pbx has a dedicated ethernet port for it. We have two trunk providers and I only see the native rtp bridge used on one of them. I will do a packet capture on the trunk interface to see if something else strange happens.
Thank you.
| Quote: | --
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161 |
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