[asterisk-users] Diagnosing call problem
Mitch Claborn
mitch_ml at claborn.net
Tue Mar 19 19:18:09 CDT 2013
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 <mitch_ml at claborn.net
> <mailto:mitch_ml at claborn.net>> 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
> <asghar144 at gmail.com <mailto:asghar144 at gmail.com>
> <mailto: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>
> <mailto: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>
> <http://sip:KWakmn@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>
> <mailto:sip%3A2392230612 at 172.__16.0.245
> <mailto:sip%253A2392230612 at 172.16.0.245>>>;__tag=as4b489afc
> To: <sip:KWakmn at 172.16.0.71:5060
> <http://sip:KWakmn@172.16.0.71:5060>
> <http://sip:KWakmn@172.16.0.__71:5060
> <http://sip:KWakmn@172.16.0.71:5060>>>
> Contact: <sip:2392230612
> <tel:2392230612>@172.16.0.245:__5060
> <http://sip:2392230612@172.16.__0.245:5060
> <http://sip:2392230612@172.16.0.245:5060>>>
> Call-ID:
> 52106f231b41169c7eabd3b43d0fc6____e8 at 172.16.0.245:5060
> <http://52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060>
>
> <http://__52106f231b41169c7eabd3b43d0fc6__e8@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
> <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://52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060>
>
> <http://__52106f231b41169c7eabd3b43d0fc6__e8@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>
> <mailto:sip%3A2392230612 at 172.__16.0.245
> <mailto:sip%253A2392230612 at 172.16.0.245>>>;__tag=as4b489afc
> To: <sip:KWakmn at 172.16.0.71
> <mailto:sip%3AKWakmn at 172.16.0.71>
> <mailto:sip%3AKWakmn at 172.16.0.__71
> <mailto:sip%253AKWakmn 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>
> <http://sip:KWakmn@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://52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060>
>
> <http://__52106f231b41169c7eabd3b43d0fc6__e8@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>
> <mailto:sip%3A2392230612 at 172.__16.0.245
> <mailto:sip%253A2392230612 at 172.16.0.245>>>;__tag=as4b489afc
> To: <sip:KWakmn at 172.16.0.71
> <mailto:sip%3AKWakmn at 172.16.0.71>
> <mailto:sip%3AKWakmn at 172.16.0.__71
> <mailto:sip%253AKWakmn 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>
> <http://sip:KWakmn@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>
> <http://sip:KWakmn@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>
> <mailto:sip%3A2392230612 at 172.__16.0.245
> <mailto:sip%253A2392230612 at 172.16.0.245>>>;__tag=as4b489afc
> To: <sip:KWakmn at 172.16.0.71:5060
> <http://sip:KWakmn@172.16.0.71:5060>
> <http://sip:KWakmn@172.16.0.__71:5060
> <http://sip:KWakmn@172.16.0.71:5060>>>;__tag=7543f39a-7ca0-__434b-8281-__e6dc2adc4aa3
> Contact: <sip:2392230612
> <tel:2392230612>@172.16.0.245:__5060
> <http://sip:2392230612@172.16.__0.245:5060
> <http://sip:2392230612@172.16.0.245:5060>>>
> Call-ID:
> 52106f231b41169c7eabd3b43d0fc6____e8 at 172.16.0.245:5060
> <http://52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060>
>
> <http://__52106f231b41169c7eabd3b43d0fc6__e8@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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