[asterisk-users] Diagnosing call problem
Mitch Claborn
mitch_ml at claborn.net
Tue Mar 19 17:57:55 CDT 2013
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 <asghar144 at gmail.com
> <mailto:asghar144 at gmail.com>> 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
> <mitch_ml at claborn.net <mailto:mitch_ml at claborn.net>> 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:KWakmn at 172.16.0.71:5060
> <http://sip:KWakmn@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
> <mailto:sip%3A2392230612 at 172.16.0.245>>;__tag=as4b489afc
> To: <sip:KWakmn at 172.16.0.71:5060
> <http://sip:KWakmn@172.16.0.71:5060>>
> Contact: <sip:2392230612 at 172.16.0.245:__5060
> <http://sip:2392230612@172.16.0.245:5060>>
> Call-ID: 52106f231b41169c7eabd3b43d0fc6__e8 at 172.16.0.245:5060
> <http://52106f231b41169c7eabd3b43d0fc6e8@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
> <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: 52106f231b41169c7eabd3b43d0fc6__e8 at 172.16.0.245:5060
> <http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060>
> From: <sip:2392230612 at 172.16.0.245
> <mailto:sip%3A2392230612 at 172.16.0.245>>;__tag=as4b489afc
> To: <sip:KWakmn at 172.16.0.71
> <mailto:sip%3AKWakmn at 172.16.0.71>>;tag=__7543f39a-7ca0-434b-8281-__e6dc2adc4aa3
> CSeq: 102 INVITE
> Contact: <sip:KWakmn at 172.16.0.71:5060
> <http://sip:KWakmn@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: 52106f231b41169c7eabd3b43d0fc6__e8 at 172.16.0.245:5060
> <http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060>
> From: <sip:2392230612 at 172.16.0.245
> <mailto:sip%3A2392230612 at 172.16.0.245>>;__tag=as4b489afc
> To: <sip:KWakmn at 172.16.0.71
> <mailto:sip%3AKWakmn 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
> <http://sip:KWakmn@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
> <http://sip:KWakmn@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
> <mailto:sip%3A2392230612 at 172.16.0.245>>;__tag=as4b489afc
> To: <sip:KWakmn at 172.16.0.71:5060
> <http://sip:KWakmn@172.16.0.71:5060>>;__tag=7543f39a-7ca0-434b-8281-__e6dc2adc4aa3
> Contact: <sip:2392230612 at 172.16.0.245:__5060
> <http://sip:2392230612@172.16.0.245:5060>>
> Call-ID: 52106f231b41169c7eabd3b43d0fc6__e8 at 172.16.0.245:5060
> <http://52106f231b41169c7eabd3b43d0fc6e8@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
>
>
>
> --
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