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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.245:5060;branch=z9hG4bK19e2246d
Max-Forwards: 70
From: <sip:[email protected]>;tag=as4b489afc
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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: [email protected]:5060
From: <sip:[email protected]>;tag=as4b489afc
To: <sip:[email protected]>;tag=7543f39a-7ca0-434b-8281-e6dc2adc4aa3
CSeq: 102 INVITE
Contact: <sip:[email protected]: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: [email protected]:5060
From: <sip:[email protected]>;tag=as4b489afc
To: <sip:[email protected]>;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:[email protected]: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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.245:5060;branch=z9hG4bK289d6da2
Max-Forwards: 70
From: <sip:[email protected]>;tag=as4b489afc
To: <sip:[email protected]:5060>;tag=7543f39a-7ca0-434b-8281-e6dc2adc4aa3
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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@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 <[email protected]
<mailto:[email protected]>> 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
<[email protected] <mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>> 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
<[email protected]
<mailto:[email protected]>
<mailto:satish4asterisk@gmail.__com
<mailto:[email protected]>>
<mailto:satish4asterisk@gmail.
<mailto:satish4asterisk@gmail.>____com
<mailto:satish4asterisk@gmail.__com
<mailto:[email protected]>>>> wrote:
On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn
<[email protected]
<mailto:[email protected]> <mailto:[email protected]
<mailto:[email protected]>>
<mailto:[email protected]
<mailto:[email protected]> <mailto:[email protected]
<mailto:[email protected]>>>__> 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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