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
--
_________________________________________________________________________
-- Bandwidth and Colocation Provided by
http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/__mailman/listinfo/asterisk-__users
<http://lists.digium.com/mailman/listinfo/asterisk-users>
--
_________________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/__mailman/listinfo/asterisk-__users
<http://lists.digium.com/mailman/listinfo/asterisk-users>
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
