[asterisk-bugs] [JIRA] (ASTERISK-26593) bridge_native_rtp: Switching to native rtp disables RTCP reporting

Luke Escude (JIRA) noreply at issues.asterisk.org
Mon Nov 14 15:48:10 CST 2016


    [ https://issues.asterisk.org/jira/browse/ASTERISK-26593?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233707#comment-233707 ] 

Luke Escude commented on ASTERISK-26593:
----------------------------------------

Trunk Configuration:

[flowroute]
username=REDACTED
fromuser=REDACTED
secret=REDACTED
host=sip.flowroute.com
type=friend
context=from-trunk
insecure=port,invite
trustrpid=yes
sendrpid=yes
directmedia=no
qualify=yes
keepalive=45
nat=force_rport,comedia
dtmfmode=rfc2833
disallow=all
allow=ulaw
t38pt_udptl=yes,redundancy,maxdatagram=400
defaultexpiry=120
registertimeout=1
registerattempts=0


Endpoint Config:
[1033]
secret=REDACTED
dial=SIP/1033
mailbox=1033 at voicemail
callerid=Luke Office <1033>
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0
disallow=all
allow=ulaw
allow=ilbc:30
allow=g729
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=pai
type=friend
nat=force_rport,comedia
port=5060
qualify=10000
qualifyfreq=15
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
namedcallgroup=
namedpickupgroup=
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

> bridge_native_rtp: Switching to native rtp disables RTCP reporting
> ------------------------------------------------------------------
>
>                 Key: ASTERISK-26593
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26593
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp
>    Affects Versions: 13.12.2
>         Environment: CentOS x64
>            Reporter: Luke Escude
>            Assignee: Unassigned
>            Severity: Minor
>         Attachments: sip_debug.rtf
>
>
> As soon as the call switches from simple_bridge to native_rtp, the server stops receiving RTCP information, and the call will drop because it thinks there's inactivity. We've had to set the rtp timeout to 24 hours to compensate for this.
> Log readout during a phone call:
> vpbx36*CLI> sip show channelstats
> Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
> 216.115.69.144   4300d26f61c  00:00:24 0000001231  0000000000 ( 0.00%) 0.0000 0000000779  0000000000 ( 0.00%) 0.0001
> 98.6.78.250      1180865772-  00:00:24 0000000779  0000000000 ( 0.00%) 0.0000 0000000785  0000000000 ( 0.00%) 0.0023
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
> 216.115.69.144   4300d26f61c  00:00:25 0000001254  0000000000 ( 0.00%) 0.0000 0000000794  0000000000 ( 0.00%) 0.0001
> 98.6.78.250      1180865772-  00:00:25 0000000794  0000000000 ( 0.00%) 0.0000 0000000800  0000000000 ( 0.00%) 0.0020
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
> 216.115.69.144   4300d26f61c  00:00:25 0000001281  0000000000 ( 0.00%) 0.0000 0000000812  0000000000 ( 0.00%) 0.0001
> 98.6.78.250      1180865772-  00:00:25 0000000812  0000000000 ( 0.00%) 0.0000 0000000818  0000000000 ( 0.00%) 0.0025
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
> 216.115.69.144   4300d26f61c  00:00:26 0000001305  0000000000 ( 0.00%) 0.0000 0000000828  0000000000 ( 0.00%) 0.0001
> 98.6.78.250      1180865772-  00:00:26 0000000828  0000000000 ( 0.00%) 0.0000 0000000834  0000000000 ( 0.00%) 0.0023
> 2 active SIP channels
>     -- SIP/flowroute-000000c3 answered SIP/236-000000c2
>     -- Channel SIP/flowroute-000000c3 joined 'simple_bridge' basic-bridge <f471fb8f-d8ae-4aef-b164-c35f20846fb3>
>     -- Channel SIP/236-000000c2 joined 'simple_bridge' basic-bridge <f471fb8f-d8ae-4aef-b164-c35f20846fb3>
>        > Bridge f471fb8f-d8ae-4aef-b164-c35f20846fb3: switching from simple_bridge technology to native_rtp
>        > Locally RTP bridged 'SIP/236-000000c2' and 'SIP/flowroute-000000c3' in stack
>        > Locally RTP bridged 'SIP/236-000000c2' and 'SIP/flowroute-000000c3' in stack
> vpbx36*CLI> sip show channelstats
> Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
> 216.115.69.144   4300d26f61c  00:00:26 0000001330  0000000000 ( 0.00%) 0.0000 0000000842  0000000000 ( 0.00%) 0.0001
> 98.6.78.250      1180865772-  00:00:26 0000000842  0000000000 ( 0.00%) 0.0000 0000000854  0000000000 ( 0.00%) 0.0026
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
> 216.115.69.144   4300d26f61c  00:00:28 0000001408  0000000000 ( 0.00%) 0.0000 0000000888  0000000000 ( 0.00%) 0.0001
> 98.6.78.250      1180865772-  00:00:28 0000000888  0000000000 ( 0.00%) 0.0000 0000000932  0000000000 ( 0.00%) 0.0026
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
> 216.115.69.144   4300d26f61c  00:00:28 0000001434  0000000000 ( 0.00%) 0.0000 0000000906  0000000000 ( 0.00%) 0.0001
> 98.6.78.250      1180865772-  00:00:28 0000000906  0000000000 ( 0.00%) 0.0000 0000000958  0000000000 ( 0.00%) 0.0026
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
> 216.115.69.144   4300d26f61c  00:00:29 0000001457  0000000000 ( 0.00%) 0.0000 0000000921  0000000000 ( 0.00%) 0.0001
> 98.6.78.250      1180865772-  00:00:29 0000000921  0000000000 ( 0.00%) 0.0000 0000000981  0000000000 ( 0.00%) 0.0026
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
> 216.115.69.144   4300d26f61c  00:00:36 0000001808  0000000000 ( 0.00%) 0.0000 0000001155  0000000000 ( 0.00%) 0.0001
> 98.6.78.250      1180865772-  00:00:36 0000001155  0000000000 ( 0.00%) 0.0000 0000001332  0000000000 ( 0.00%) 0.0026
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
> 216.115.69.144   4300d26f61c  00:00:37 0000001845  0000000000 ( 0.00%) 0.0000 0000001180  0000000000 ( 0.00%) 0.0001
> 98.6.78.250      1180865772-  00:00:37 0000001180  0000000000 ( 0.00%) 0.0000 0000001369  0000000000 ( 0.00%) 0.0026



--
This message was sent by Atlassian JIRA
(v6.2#6252)



More information about the asterisk-bugs mailing list