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Summary:ASTERISK-13638: Missing one way audio on one phone after the bridge
Reporter:Drake Swokowski (drake)Labels:
Date Opened:2009-02-24 09:16:23.000-0600Date Closed:2009-06-08 11:11:26
Priority:BlockerRegression?No
Status:Closed/CompleteComponents:Core/RTP
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 14539_rtp_trace_screenshot.JPG
( 1) 14539_verbose_debug.txt
Description:Asterisk should not drop the new rtp stream with time stamp zero after bridging two sip peers.

Details:
SIP phone A and B are registered to *.

A SIP call comes from a cisco pstn gateway to * on extension 2000, which bridges  to Phone B.

After bridging, there is no audio in Phone B.

The party that sent the call through the gateway, can both receive and send audio.

From the sip traffic capture, it seems that the incoming rtp from the gateway has no marker bit set and has time stamp =0. * is dropping this rtp packet as it isn't forwarded to the phone. Rather it forwards the next rtp packet with time stamped preserved (ts = 160) after setting its marker bit.






****** ADDITIONAL INFORMATION ******

Just for your information, after above bridge which has one way audio loss,if the sip phone A makes a call to another extension 3001, it bridges it with the phone B. The phone B now has audio back.

seen in 1.6.0.5 and 1.6.0.6-rc1
Comments:By: Drake Swokowski (drake) 2009-02-24 09:18:33.000-0600

extensions.conf and sip.conf respectively
[default]
include => inbound
;exten => _00XXXXXXXXX,1,Dial(2010/${EXTEN:1})
;exten => 2010,1,Dial(2010/${EXTEN:1})
exten => s,1,Hangup();

[macro-storec2]
exten => s,1,Set(DB(ch/c2)=${CDR(channel)})
exten => s,2,Set(DB(ch/${ARG1})=${CDR(channel)})
exten => s,3,NoOp(OPTI_DEBUG In Macro, c2 value: ${DB(ch/c2)}, ARG1: ${ARG1})

[inbound]
;exten => 0390xxxxxx,1,Dial(SIP/101)

exten => 2010,1,Set(DB(ch/c1)=${CDR(channel)})
exten => 2010,2,NoOp()
exten => 2010,3,Dial(SIP/2010,,M(storec2^${EXTEN}))

exten => 2000,1,Set(DB(ch/g1)=${CDR(channel)});bridge this gsm call channel
exten => 2000,2,Set(VCCCHAN1=${DB(ch/c2)})
exten => 2000,3,NoOp(####################HO2GSM Callerid: ${CALLERID(num)} and cdr channel: ${CDR(channel)} ###)
exten => 2000,4,Bridge(${VCCCHAN1})
exten => 2000,5,NoOp(################### Bridge Result: ${BRIDGERESULT}#################)

;ho2voip
;exten => 2001,1,Set(VCCCHAN1=${DB(ch/g1)})

exten => 2001,1,Set(DB(ch/v1loop)=${CDR(channel)})
exten => 2001,2,NoOp(####################VDINFO Callerid: ${CALLERID(num)} and channel: ${CDR(channel)}and VCCCHAN1  ${VCCCHAN1} ###)
exten => 2001,3,Dial(SIP/2011,,G(4)) ; dial ho initiator ie aparty
exten => 2001,4,Set(DB(ch/v1)=${CDR(channel)})
exten => 2001,5,NoOp();bridge current call channel with given gsm channel g1
exten => 2001,6,Bridge(${DB(ch/c2)});
exten => 2001,7,NoOp(################### Bridge Result: ${BRIDGERESULT}#################)

---------------------------------------------

[general]
promiscredir=no
tcpenable=no
bindport=5060
srvlookup=no
rtpkeepalive=1
;following must be greater than rtptimeout
;rtpholdtimeout=
;rtptimeout
context=default
permit=0.0.0.0
disallow=all
allow=alaw
insecure=port,invite
[2010]
host=dynamic
context=inbound
canreinvite=no
type=friend
nat=yes
allow=alaw
;qualify=1000                    ; Consider it down if it's 1 second to reply
; qualify=yes; uses default value
;qualifyfreq=60

[2011]
host=dynamic
context=inbound
canreinvite=no
type=friend
nat=yes
allow=alaw
;qualify=1000                    ; Consider it down if it's 1 second to reply
; qualify=yes; uses default value
;qualifyfreq=60
By: Drake Swokowski (drake) 2009-02-24 09:57:01.000-0600

I couldn't extract the wireshark trace so I sent the screenshot.
By: Joshua C. Colp (jcolp) 2009-02-25 09:20:23.000-0600

I've looked into this and have a thought - can you please try commenting out rtpkeepalive in your sip.conf and seeing if this changes things?
By: Joshua C. Colp (jcolp) 2009-03-02 13:15:23.000-0600

Per jeremy_g on IRC disabling the option did not work.
By: Leif Madsen (lmadsen) 2009-05-07 08:24:33

Are you still having this issue with latest 1.6.0.x release?
By: Leif Madsen (lmadsen) 2009-05-22 12:37:06

Closed due to lack of response.