Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
Andres wrote: I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an idea about how to best go about troubleshooting this? Use tcpdump to capture to a file both call scenarios. Then use Wireshark to open the file. You can then do an 'RTP- Show All Streams' Analysis of the calls. That alone would reveal whether the Audio is really there or not. You can export that G711 Payload and listen to it with the Windows Media Player. I'm running wireshark 1.0.3. I've opened the captures... How do I examine the streams? I don't follow what you're saying above. And does anyone have a plugin that would allow actual playback of the .pcap files' audio packets? Thanks, -Philip If you don't see the RTP in one direction then you might have a signalling problem. Andres http://www.neuroredes.com Thanks, -Philip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
Philip Prindeville wrote:
Well, things just got a lot more interesting... Adding Monitor() to an
extension ends the one-way voice problem on inbound calls!
So an incoming call gets handled as:
[ctc-incoming]
exten = 208345,1,Noop()
exten = 208345,n,Log(NOTICE: RDNIS: ${CALLERID(rdnis)} ANI:
${CALLERID(ani)})
exten = 208345,n,Goto(redfish-pstn,s,1)
...
[redfish-pstn]
exten = s,1(incoming),Noop()
exten = s,n,Answer()
exten = s,n,Wait(0.5)
...
some filters for bogus ANI's like 8 goes to badani below
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,nWaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
exten = s,n(badani),Log(DEBUG,ANI: ${CALLERID(ani)} clearing)
exten = s,n,Playback(privacy-unident)
exten = s,n,Wait(0.5)
exten = s,n,Congestion()
exten = s,n,Hangup()
include = redfish-extens
exten = i,1,NoOp(Invalid: ${EXTEN})
exten = i,n,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
[redfish-extens]
...
exten = 113,1,Monitor(wav,,w); for debugging
exten = 113,n,Macro(stdexten,113,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
...
exten = 113,1,Macro(stdexten,119,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
Err, sorry. Typo. That was:
exten = 119,1,Macro(stdexten,119,${GUEST},redfish)
exten = 119,n,Goto(s,exten)
-Philip
So I don't get this at all. If I dial 208345, then enter '119' as
the extension, it rings on a few phones (including a Xlite softphone)
and if I pick up on any of those, I get one-way voice (I can hear the
caller but they can't hear me).
If I enter '113' as the extension, it rings on two SPA-942's (one of
which is the same as above, just a different line presentation)... and
if I answer, then I get two-way voice! Only difference is the Monitor()
statement.
I'm starting to suspect it's a CODEC issue in Asterisk, though (a) why
Asterisk would need to transcode a call between two uLaw endpoints, I
don't know... and (b) why is it staying in the Media path at all?
I have the SIP peer that the calls come in on as:
[sip-proxy]
...
type=peer
nat=no
canreinvite=no
reinvite=no
Anyone know why the Monitor() would change the duplex(ity) of the audio
stream? I'm baffled (no pun intended). And is there any debugging I
can turn on to reveal CODEC behavior that might differ from 113 and 119?
Thanks,
-Philip
___
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Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
A similar issue happens to us. Make sure that, for inbound AND outbound calls rtp packets are reaching the other endpoint. If a NAT device(s) is between the endpoints make sure that the device NATs the traffic on BOTH ways (inbound AND outbound). Regards On Saturday 27 September 2008 23:54:37 Philip Prindeville wrote: I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of the RTP traffic, and the addresses and port numbers correspond to what's in the SIP INVITE/OK messages (assuming that they don't somehow get munged by NAT after tcpdump looks at them -- there is no NAT device upstream of my Asterisk firewall). I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an idea about how to best go about troubleshooting this? Thanks, -Philip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
On Sat, Sep 27, 2008 at 5:54 PM, Philip Prindeville [EMAIL PROTECTED] wrote: I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of the RTP traffic, and the addresses and port numbers correspond to what's in the SIP INVITE/OK messages (assuming that they don't somehow get munged by NAT after tcpdump looks at them -- there is no NAT device upstream of my Asterisk firewall). I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an idea about how to best go about troubleshooting this? Thanks, -Philip Philip, We were recently having a few call quality problems with one of our carriers, including very mysterious one way audio on specific pieces of equipment. I created a call generator using Playtones/Monitor to record all four audio paths (only two are important) of a successful call and analyze the resulting recordings with ecasound to detect distortion, one-way audio, audio drops, etc. After several hundred calls we were able to get the carrier to correct the offending pieces of equipment. I'm looking into a way to do this in real time but for now this collection of scripts works pretty well. It's not ready for release but I could get it to you shortly for some testing. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
Well, things just got a lot more interesting... Adding Monitor() to an
extension ends the one-way voice problem on inbound calls!
So an incoming call gets handled as:
[ctc-incoming]
exten = 208345,1,Noop()
exten = 208345,n,Log(NOTICE: RDNIS: ${CALLERID(rdnis)} ANI:
${CALLERID(ani)})
exten = 208345,n,Goto(redfish-pstn,s,1)
...
[redfish-pstn]
exten = s,1(incoming),Noop()
exten = s,n,Answer()
exten = s,n,Wait(0.5)
...
some filters for bogus ANI's like 8 goes to badani below
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,nWaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
exten = s,n(badani),Log(DEBUG,ANI: ${CALLERID(ani)} clearing)
exten = s,n,Playback(privacy-unident)
exten = s,n,Wait(0.5)
exten = s,n,Congestion()
exten = s,n,Hangup()
include = redfish-extens
exten = i,1,NoOp(Invalid: ${EXTEN})
exten = i,n,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
[redfish-extens]
...
exten = 113,1,Monitor(wav,,w); for debugging
exten = 113,n,Macro(stdexten,113,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
...
exten = 113,1,Macro(stdexten,119,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
So I don't get this at all. If I dial 208345, then enter '119' as
the extension, it rings on a few phones (including a Xlite softphone)
and if I pick up on any of those, I get one-way voice (I can hear the
caller but they can't hear me).
If I enter '113' as the extension, it rings on two SPA-942's (one of
which is the same as above, just a different line presentation)... and
if I answer, then I get two-way voice! Only difference is the Monitor()
statement.
I'm starting to suspect it's a CODEC issue in Asterisk, though (a) why
Asterisk would need to transcode a call between two uLaw endpoints, I
don't know... and (b) why is it staying in the Media path at all?
I have the SIP peer that the calls come in on as:
[sip-proxy]
...
type=peer
nat=no
canreinvite=no
reinvite=no
Anyone know why the Monitor() would change the duplex(ity) of the audio
stream? I'm baffled (no pun intended). And is there any debugging I
can turn on to reveal CODEC behavior that might differ from 113 and 119?
Thanks,
-Philip
Philip Prindeville wrote:
I've got the following situation. I'm running Asterisk 1.4.18 on a
firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones
behind it.
I'm peering SIP with a Coppercom switch sitting behind an SBC.
On outbound calls, I get 2-way voice, no worries.
On inbound calls, I get one-way voice (I can hear the caller but they
can't hear me).
I've looked at tcpdumps of the RTP traffic, and the addresses and port
numbers correspond to what's in the SIP INVITE/OK messages (assuming
that they don't somehow get munged by NAT after tcpdump looks at them --
there is no NAT device upstream of my Asterisk firewall).
I'll look into using Record() or Monitor() to capture the phone call,
but if there's any conversion being done by codecs then that won't
eliminate the possibility that the code itself is misconfigured or buggy
and generating a bad stream on one of the legs...
Anyone have an idea about how to best go about troubleshooting this?
Thanks,
-Philip
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an idea about how to best go about troubleshooting this? Use tcpdump to capture to a file both call scenarios. Then use Wireshark to open the file. You can then do an 'RTP- Show All Streams' Analysis of the calls. That alone would reveal whether the Audio is really there or not. You can export that G711 Payload and listen to it with the Windows Media Player. If you don't see the RTP in one direction then you might have a signalling problem. Andres http://www.neuroredes.com Thanks, -Philip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of the RTP traffic, and the addresses and port numbers correspond to what's in the SIP INVITE/OK messages (assuming that they don't somehow get munged by NAT after tcpdump looks at them -- there is no NAT device upstream of my Asterisk firewall). I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an idea about how to best go about troubleshooting this? Thanks, -Philip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
