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

[Rusty Newton (JIRA)]

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

Rusty Newton edited comment on ASTERISK-26593 at 11/14/16 4:38 PM:
-------------------------------------------------------------------

Trunk Configuration:
{noformat}
[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
{noformat}

Endpoint Config:
{noformat}
[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
{noformat}


was (Author: lukeescude):
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: console_log.txt, 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:
> [Edit by Rusty - moved debug to console_log.txt]



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