Some Calls Drop After 30 Seconds

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Asterisk Users 3 Comments

    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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3 thoughts on - Some Calls Drop After 30 Seconds

  • 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.

  • Hi Carlos

    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?

    Not all, just some. After looking at the log files the only

  •     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.