That change did not fix the problem. Below is another trace from a
failed call this morning. 172.16.0.71 is the client, 172.16.0.245 is
the Asterisk server. All the RTP packets after the SIP are from server
to client.
Any further ideas are appreciated. (If I don't get this fixed this
week, I won't get to go home on Friday!)
-----------------------------------------------
No. Time Source Destination
Protocol Length Info
4528 12:14:07.219165 172.16.0.245 172.16.0.71
SIP/SDP 910 Request: INVITE sip:[email protected]:5060, with
session description
Frame 4528: 910 bytes on wire (7280 bits), 910 bytes captured (7280 bits)
Ethernet II, Src: 90:b1:1c:0d:c4:35 (90:b1:1c:0d:c4:35), Dst:
Dell_e7:fc:b0 (00:25:64:e7:fc:b0)
Internet Protocol Version 4, Src: 172.16.0.245 (172.16.0.245), Dst:
172.16.0.71 (172.16.0.71)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: INVITE sip:[email protected]:5060 SIP/2.0
Message Header
Via: SIP/2.0/UDP 172.16.0.245:5060;branch=z9hG4bK7e5fc96a
Max-Forwards: 70
From: <sip:[email protected]>;tag=as5a63ac9a
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: Wed, 20 Mar 2013 17:14:07 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
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 582679053 582679053 IN
IP4 172.16.0.245
Session Name (s): Asterisk PBX 11.1.0
Connection Information (c): IN IP4 172.16.0.245
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 28340
RTP/AVP 0 8 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): ptime:20
Media Attribute (a): sendrecv
------------------------------------------
No. Time Source Destination
Protocol Length Info
1 12:14:07.251118 172.16.0.71 172.16.0.245 SIP
542 Status: 180 Ringing
Frame 1: 542 bytes on wire (4336 bits), 542 bytes captured (4336 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: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Status-Line: SIP/2.0 180 Ringing
Message Header
Via: SIP/2.0/UDP
172.16.0.245:5060;received=172.16.0.245;branch=z9hG4bK7e5fc96a
Call-ID: [email protected]:5060
From: <sip:[email protected]>;tag=as5a63ac9a
To:
<sip:[email protected]>;tag=923b9add-5ef2-4f6f-a3f5-4627109e079c
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
------------------------------------------
No. Time Source Destination
Protocol Length Info
5 12:14:18.055112 172.16.0.71 172.16.0.245
SIP/SDP 834 Status: 200 OK, with session description
Frame 5: 834 bytes on wire (6672 bits), 834 bytes captured (6672 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: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP
172.16.0.245:5060;received=172.16.0.245;branch=z9hG4bK7e5fc96a
Call-ID: [email protected]:5060
From: <sip:[email protected]>;tag=as5a63ac9a
To:
<sip:[email protected]>;tag=923b9add-5ef2-4f6f-a3f5-4627109e079c
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
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): asset071 3572788447 1 IN IP4
172.16.0.71
Session Name (s): sflphone
Connection Information (c): IN IP4 172.16.0.71
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 45208 RTP/AVP 0
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): sendrecv
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): rtcp:45209 IN IP4 172.16.0.71
----------------------------------
No. Time Source Destination
Protocol Length Info
6 12:14:18.056116 172.16.0.245 172.16.0.71 SIP
463 Request: ACK sip:[email protected]:5060
Frame 6: 463 bytes on wire (3704 bits), 463 bytes captured (3704 bits)
Ethernet II, Src: 90:b1:1c:0d:c4:35 (90:b1:1c:0d:c4:35), Dst:
Dell_e7:fc:b0 (00:25:64:e7:fc:b0)
Internet Protocol Version 4, Src: 172.16.0.245 (172.16.0.245), Dst:
172.16.0.71 (172.16.0.71)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: ACK sip:[email protected]:5060 SIP/2.0
Message Header
Via: SIP/2.0/UDP 172.16.0.245:5060;branch=z9hG4bK5a4eafc5
Max-Forwards: 70
From: <sip:[email protected]>;tag=as5a63ac9a
To:
<sip:[email protected]:5060>;tag=923b9add-5ef2-4f6f-a3f5-4627109e079c
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.1.0
Content-Length: 0
----------
Mitch
On 03/19/2013 07:18 PM, Mitch Claborn wrote:
Good point. I changed to 10000 - 40000.
Mitch
On 03/19/2013 06:17 PM, Asghar Mohammad wrote:
i had this problem with a gateway witch was configured from 1000 to 3000
and some time he was using ports above 2000 and result was one way voice
rtp port range is where asterisk expect audio, you should not use ports
below 10000 because they are in use of other services like 5060 for sip.
On Tue, Mar 19, 2013 at 11:57 PM, Mitch Claborn <[email protected]
<mailto:[email protected]>> wrote:
This was the client sending from port 39409 to server port 13429,
which is in the range. From what I read, the rtpstart and rtpend
define the range that is available for use on the server, so I'm not
sure this will apply.
But, I've set my range to 5000 - 40000. I'll find out tomorrow if
it makes any difference.
Where is a good place to find documentation on the various fields in
the INVITE SIP message and the response? I'd like to dig into them
and try to understand them more completely.
Mitch
On 03/19/2013 05:02 PM, Asghar Mohammad wrote:
hi,
"User Datagram Protocol, Src Port: 39409 (39409), Dst Port:
13429 (13429)"
copy from asterisk 11 rtp.conf
rtpstart=10000
rtpend=20000
have you changed port range? if no then
your client sending rtp to a port higher then configured in
rtp port
range and asterisk ignore that port.
try to change rtpend=30000 or if there is option in
softphone restrict it to use same range as in rtp.conf.
let me know if this solve you problem.
On Tue, Mar 19, 2013 at 10:54 PM, Asghar Mohammad
<[email protected] <mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>> wrote:
hi,
"User Datagram Protocol, Src Port: 39409 (39409), Dst Port:
13429
(13429)"
copy from asterisk 11 rtp.conf
rtpstart=10000
rtpend=20000
have you changed port range? if no then
your client sending rtp to a port higher then configured in
rtp port
range and asterisk ignore that port.
try to change rtpend=30000 or if there is option in
softphone restrict it to use same range as in rtp.conf.
let me know if this solve you problem.
On Tue, Mar 19, 2013 at 10:22 PM, Mitch Claborn
<[email protected] <mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>>
wrote:
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
<http://sip:[email protected]:5060>
<http://sip:[email protected].__71:5060
<http://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]
<mailto:sip%[email protected]>
<mailto:sip%3A2392230612@172.__16.0.245
<mailto:sip%[email protected]>>>;__tag=as4b489afc
To: <sip:[email protected]:5060
<http://sip:[email protected]:5060>
<http://sip:[email protected].__71:5060
<http://sip:[email protected]:5060>>>
Contact: <sip:2392230612
<tel:2392230612>@172.16.0.245:__5060
<http://sip:[email protected].__0.245:5060
<http://sip:[email protected]:5060>>>
Call-ID:
[email protected]:5060
<http://[email protected]:5060>
<http://[email protected]:5060
<http://[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
<http://www.mcmurrayhatchery.__com
<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
<http://[email protected]:5060>
<http://[email protected]:5060
<http://[email protected]:5060>>
From: <sip:[email protected]
<mailto:sip%[email protected]>
<mailto:sip%3A2392230612@172.__16.0.245
<mailto:sip%[email protected]>>>;__tag=as4b489afc
To: <sip:[email protected]
<mailto:sip%[email protected]>
<mailto:sip%[email protected].__71
<mailto:sip%[email protected]>>>;tag=__7543f39a-7ca0-434b-__8281-__e6dc2adc4aa3
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060
<http://sip:[email protected]:5060>
<http://sip:[email protected].__71:5060
<http://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
<http://[email protected]:5060>
<http://[email protected]:5060
<http://[email protected]:5060>>
From: <sip:[email protected]
<mailto:sip%[email protected]>
<mailto:sip%3A2392230612@172.__16.0.245
<mailto:sip%[email protected]>>>;__tag=as4b489afc
To: <sip:[email protected]
<mailto:sip%[email protected]>
<mailto:sip%[email protected].__71
<mailto: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
<http://sip:[email protected]:5060>
<http://sip:[email protected].__71:5060
<http://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
<http://sip:[email protected]:5060>
<http://sip:[email protected].__71:5060
<http://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]
<mailto:sip%[email protected]>
<mailto:sip%3A2392230612@172.__16.0.245
<mailto:sip%[email protected]>>>;__tag=as4b489afc
To: <sip:[email protected]:5060
<http://sip:[email protected]:5060>
<http://sip:[email protected].__71:5060
<http://sip:[email protected]:5060>>>;__tag=7543f39a-7ca0-__434b-8281-__e6dc2adc4aa3
Contact: <sip:2392230612
<tel:2392230612>@172.16.0.245:__5060
<http://sip:[email protected].__0.245:5060
<http://sip:[email protected]:5060>>>
Call-ID:
[email protected]:5060
<http://[email protected]:5060>
<http://[email protected]:5060
<http://[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
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