[asterisk-users] Diagnosing call problem
Mitch Claborn
mitch_ml at claborn.net
Tue Mar 19 16:22:13 CDT 2013
We have Ubuntu 12.04 clients, using either SFLPhone or Bria 3.
There is no NAT involved in the network at all (it is disabled in sip.conf).
Here are the SIP messages capture via wireshark on the client during one
problem call. Following these SIP messages, the wireshark trace shows
only RTP packets from server (172.16.0.245) to client (172.16.0.71)
except for an occasional RTCP packet from client to server (sample below).
Any help is appreciated. The uses are really beating me up to get this
fixed.
--------------------
INVITE sip:KWakmn at 172.16.0.71:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.245:5060;branch=z9hG4bK19e2246d
Max-Forwards: 70
From: <sip:2392230612 at 172.16.0.245>;tag=as4b489afc
To: <sip:KWakmn at 172.16.0.71:5060>
Contact: <sip:2392230612 at 172.16.0.245:5060>
Call-ID: 52106f231b41169c7eabd3b43d0fc6e8 at 172.16.0.245:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.1.0
Date: Tue, 19 Mar 2013 20:47:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
X-mm-call: http://www.mcmurrayhatchery.com
Content-Type: application/sdp
Content-Length: 257
v=0
o=root 682517197 682517197 IN IP4 172.16.0.245
s=Asterisk PBX 11.1.0
c=IN IP4 172.16.0.245
t=0 0
m=audio 13428 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
-------------------------------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
172.16.0.245:5060;received=172.16.0.245;branch=z9hG4bK19e2246d
Call-ID: 52106f231b41169c7eabd3b43d0fc6e8 at 172.16.0.245:5060
From: <sip:2392230612 at 172.16.0.245>;tag=as4b489afc
To: <sip:KWakmn at 172.16.0.71>;tag=7543f39a-7ca0-434b-8281-e6dc2adc4aa3
CSeq: 102 INVITE
Contact: <sip:KWakmn at 172.16.0.71:5060>
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL,
UPDATE, INFO, REGISTER, OPTIONS, MESSAGE, INVITE, ACK, BYE, CANCEL
Content-Length: 0
-----------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.16.0.245:5060;received=172.16.0.245;branch=z9hG4bK19e2246d
Call-ID: 52106f231b41169c7eabd3b43d0fc6e8 at 172.16.0.245:5060
From: <sip:2392230612 at 172.16.0.245>;tag=as4b489afc
To: <sip:KWakmn at 172.16.0.71>;tag=7543f39a-7ca0-434b-8281-e6dc2adc4aa3
CSeq: 102 INVITE
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL,
UPDATE, INFO, REGISTER, OPTIONS, MESSAGE, INVITE, ACK, BYE, CANCEL
Contact: <sip:KWakmn at 172.16.0.71:5060>
Supported: replaces, 100rel
Content-Type: application/sdp
Content-Length: 234
v=0
o=asset071 3572714846 1 IN IP4 172.16.0.71
s=sflphone
c=IN IP4 172.16.0.71
t=0 0
m=audio 39408 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:39409 IN IP4 172.16.0.71
-----------------------------------------------
ACK sip:KWakmn at 172.16.0.71:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.245:5060;branch=z9hG4bK289d6da2
Max-Forwards: 70
From: <sip:2392230612 at 172.16.0.245>;tag=as4b489afc
To: <sip:KWakmn at 172.16.0.71:5060>;tag=7543f39a-7ca0-434b-8281-e6dc2adc4aa3
Contact: <sip:2392230612 at 172.16.0.245:5060>
Call-ID: 52106f231b41169c7eabd3b43d0fc6e8 at 172.16.0.245:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.1.0
Content-Length: 0
------------------------------------------------------------
SAMPLE RTCP packet from client to server
No. Time Source Destination
Protocol Length Info
240 15:47:39.965483 172.16.0.71 172.16.0.245
RTCP 102 Receiver Report Source description
Frame 240: 102 bytes on wire (816 bits), 102 bytes captured (816 bits)
Ethernet II, Src: Dell_e7:fc:b0 (00:25:64:e7:fc:b0), Dst:
90:b1:1c:0d:c4:35 (90:b1:1c:0d:c4:35)
Internet Protocol Version 4, Src: 172.16.0.71 (172.16.0.71), Dst:
172.16.0.245 (172.16.0.245)
User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429)
Real-time Transport Control Protocol (Receiver Report)
[Stream setup by SDP (frame 36)]
[Setup frame: 36]
[Setup Method: SDP]
10.. .... = Version: RFC 1889 Version (2)
..0. .... = Padding: False
...0 0001 = Reception report count: 1
Packet type: Receiver Report (201)
Length: 7 (32 bytes)
Sender SSRC: 0x841ef2ea (2216620778)
Source 1
Identifier: 0x28bcc3a6 (683459494)
SSRC contents
Fraction lost: 254 / 256
Cumulative number of packets lost: 37134
Extended highest sequence number received: 37331
Sequence number cycles count: 0
Highest sequence number received: 37331
Interarrival jitter: 160008128
Last SR timestamp: 0 (0x00000000)
Delay since last SR timestamp: 0 (0 milliseconds)
Real-time Transport Control Protocol (Source description)
[Stream setup by SDP (frame 36)]
[Setup frame: 36]
[Setup Method: SDP]
10.. .... = Version: RFC 1889 Version (2)
..0. .... = Padding: False
...0 0001 = Source count: 1
Packet type: Source description (202)
Length: 6 (28 bytes)
Chunk 1, SSRC/CSRC 0x841EF2EA
Identifier: 0x841ef2ea (2216620778)
SDES items
Type: CNAME (user and domain) (1)
Length: 17
Text: kristin at localhost
Type: END (0)
[RTCP frame length check: OK - 60 bytes]
Mitch
On 03/19/2013 12:02 PM, Asghar Mohammad wrote:
> witch softphone you are using? on client pc installed some kind of
> virtualpc like vmware or virtualbox? client pc have more then one
> network interfaces?
> you can capture sip invites from soft phone by enabling debug on client
> ip sip set debug ip "ip of softphon" upload sip trace then somebody can
> halp you, should provide more information's.
>
> On Tue, Mar 19, 2013 at 5:39 PM, Mitch Claborn <mitch_ml at claborn.net
> <mailto:mitch_ml at claborn.net>> wrote:
>
> rtp debug on the calls that do not work correctly shows packets from
> server to client only, none from client to server.
>
> I do have
>
> nat=no
> directmedia=no
>
> in sip.conf. Are there other settings that might apply?
>
> This last instance that I looked at, the problem persisted even
> after restarting the client softphone program. It was fixed after
> rebooting the client computer.
>
> Any ideas on a next step for debugging? I was thinking I would
> start a wireshark trace to see if the rtp packets are actually
> leaving the client computer.
>
>
>
> Mitch
>
>
> On 03/19/2013 08:28 AM, Bharat Lalcheta wrote:
>
> rtp set debug ip 1.2.3.4
> where 1.2.3.4 is ip of your particular agent.
> Say your x agent is not getting voice, rtp debu his ip.
> You got rtp packet from and to for that ip. If you find rtp
> packet from
> your agent to your server ip and rtp packet from your server to
> agent
> ip, then no need to check anything in asterisk. Its related to your
> agent pc problem
> If you find any single side rtp, then its problem related to nat or
> direct media etc.
> if mix monitor is on storage than only you can face problem and
> thats
> also very rare. In that case you get voice in break, but it will
> be from
> both side not in single side. So, this is not your problem at all.
> Hope you will get something in rtp debug.
> R u using any trunk then also check rtp debug between your
> server and trunk
> regards,
>
> Bharat Lalcheta
>
>
> On Tue, Mar 19, 2013 at 6:39 PM, Mitch Claborn
> <mitch_ml at claborn.net <mailto:mitch_ml at claborn.net>
> <mailto:mitch_ml at claborn.net <mailto:mitch_ml at claborn.net>>> wrote:
>
> Thanks for the suggestions.
>
> 1) directmedia was taking the default of "yes". I set to "no".
> Will watch and see.
>
> 2) NAT is turned off (nat=no). I've never done any RTP
> debugging.
> Is that "rtp set debug on ip 1.2.3.4"? How would I
> interpret the
> output?
>
> 3) mixmonitor recordings are stored on a local disk (RAID
> array,
> very fast)
>
> 4) This would have to be a last resort option, as there is a
> business requirement to record the agent calls
>
>
> Mitch
>
> On 03/19/2013 12:01 AM, Bharat Lalcheta wrote:
>
> 1) Check directmedia option in sip. If enabled set it to no
> 2) Check NAT option and RTP debug in live scenario for any
> particular agent
> 3) if not solved yet, Where are your storing your
> mixmonitor
> recording?
> On any storage ? If yes, try to record on local harddisk.
> 4) Remove mixmonitor and test again
> Hope you find can find problem 99% in above scenario.
> Regards,
> Bharat Lalcheta
>
> On Tue, Mar 19, 2013 at 10:21 AM, Satish Barot
> <satish4asterisk at gmail.com
> <mailto:satish4asterisk at gmail.com>
> <mailto:satish4asterisk at gmail.__com
> <mailto:satish4asterisk at gmail.com>>
> <mailto:satish4asterisk at gmail.
> <mailto:satish4asterisk at gmail.>____com
>
> <mailto:satish4asterisk at gmail.__com
> <mailto:satish4asterisk at gmail.com>>>> wrote:
>
>
> On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn
> <mitch_ml at claborn.net
> <mailto:mitch_ml at claborn.net> <mailto:mitch_ml at claborn.net
> <mailto:mitch_ml at claborn.net>>
> <mailto:mitch_ml at claborn.net
> <mailto:mitch_ml at claborn.net> <mailto:mitch_ml at claborn.net
> <mailto:mitch_ml at claborn.net>>>__> wrote:
>
> Asterisk 11.1.0
> Various soft-phone SIP clients
> call center with 10-12 agents online at once using
> asterisk queue
>
> Occasionally an agent will get a call (or more
> often a
> series of
> calls in a row) where neither party can hear
> the other,
> or can
> only hear each other sporadically. A MixMonitor
> recording of
> the call plays only the caller - none of the
> agent's
> audio is
> heard in the recording.
>
> Looking for ideas on how to begin to diagnose
> this or clues
> about what might be wrong.
> Is there a console command that will show
> details of a
> specific
> call in progress that might have some clues?
>
> --
>
> Mitch
>
>
> --
>
>
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>
> Silly guess, If there is no then NAT did you check
> that your
> headphones work properly every time you start the
> softphone? This
> has happened to me in past.
>
> --Satish Barot
> Ahmedabad, India.
>
> --
>
>
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> --
> Bharat Lalcheta
>
>
>
>
> --
>
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> --
> Bharat Lalcheta
>
>
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