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Summary:ASTERISK-02410: compatability problem with huawei sip interface.
Reporter:mustdie (mustdie)Labels:
Date Opened:2004-09-15 11:11:26Date Closed:2011-06-07 14:10:37
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I'm dialing just fine, but i get no audio in any direction.
I get an warning in CLI: Sep 15 11:19:40 WARNING[1142106560]: chan_sip.c:2627 process_sdp: Insufficient information for SDP (m = 'audio 0 RTP/AVP 0 8', c = '')

****** ADDITIONAL INFORMATION ******

Please see session debug:

INVITE sip:19177230306@69.18.206.8 SIP/2.0
Via: SIP/2.0/UDP 216.128.80.41:5060;branch=z9hG4bK46136384
From: "7184498690" <sip:7184498690@216.128.80.41>;tag=as38627efb
To: <sip:19177230306@69.18.206.8>
Contact: <sip:7184498690@216.128.80.41>
Call-ID: 244a825041eb7f5f42495da20c7d0f25@216.128.80.41
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 15 Sep 2004 16:19:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 863 863 IN IP4 216.128.80.41
s=session
c=IN IP4 216.128.80.41
t=0 0
m=audio 19262 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=silenceSupp:off - - - -
(no NAT) to 69.18.206.8:5060
   -- Called 19177230306@mkhoban
   -- SIP/mkhoban-0360 is ringing
Sep 15 11:19:40 WARNING[1142106560]: chan_sip.c:2627 process_sdp: Insufficient information for SDP (m = 'audio 0 RTP/AVP 0 8', c = '')
list_route: hop: <sip:19177230306@69.18.206.8:5061;user=phone>
set_destination: Parsing <sip:19177230306@69.18.206.8:5061;user=phone> for address/port to send to
set_destination: set destination to 69.18.206.8, port 5061
Comments:By: twisted (twisted) 2004-09-15 11:36:06

this looks like our outbound invite, what about the incoming one?
By: Mark Spencer (markster) 2004-09-15 13:45:43

Obviously we need the other side to proceed.
By: mustdie (mustdie) 2004-09-15 13:59:20

Message from Huawei
INVITE sip:+13472201226@216.128.80.41;user=phone SIP/2.0
From: <sip:8663078072@69.18.206.8;user=phone>;tag=a5236230
To: <sip:+13472201226@216.128.80.41;user=phone>
CSeq: 1 INVITE
Call-ID: 0c5ba14004a883adc17a5502518368fc@69.18.206.8
Via: SIP/2.0/UDP 69.18.206.8:5061;branch=z9hG4bKb14240180
Contact: <sip:8663078072@69.18.206.8:5061;user=phone>
Supported: 100rel
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER
Content-Length: 242
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 122 122 IN IP4 69.18.206.8
s=Sip Call
c=IN IP4 69.18.206.221
t=0 0
m=audio 50000 RTP/AVP 0 8 4 18 2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000

Error message
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 69.18.206.8:5061;branch=z9hG4bKb14240180
From: <sip:8663078072@69.18.206.8;user=phone>;tag=a5236230
To: <sip:+13472201226@216.128.80.41;user=phone>;tag=as7be430b8
Call-ID: 0c5ba14004a883adc17a5502518368fc@69.18.206.8
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:+13472201226@216.128.80.41>
Content-Length: 0
By: Mark Spencer (markster) 2004-09-15 14:30:36

Okay this invite is failing because there is a "+" and no matching "+" in your dialplan.  no brainer.  This is not the same as your SDP problem.  To diagnose the SDP problem we need the *response* to the *INVITE* that Asterisk sends.
By: twisted (twisted) 2004-09-15 23:29:05

I got into MustDie's box and took a look at this in ethereal... here's the results:

Session Initiation Protocol
   Status line: SIP/2.0 180 Ringing
   Message Header
       From: "3472201228" <sip:3472201228@216.128.80.41>;tag=as1b5cd2e7
       To: <sip:19177230306@69.18.206.8>;tag=b7c53fe6
       CSeq: 102 INVITE
       Call-ID: 789697e678776aab545459c817d8aaf3@216.128.80.41
       Via: SIP/2.0/UDP 216.128.80.41:5060;branch=z9hG4bK7dea1a59
       Contact: <sip:19177230306@69.18.206.8:5061;user=phone>
       Content-Length: 103
       Content-Type: application/sdp
Session Description Protocol
   Session Description Protocol Version (v): 0
   Owner/Creator, Session Id (o): HuaweiSoftX3000 65 65 IN IP4 69.18.206.8
       Owner Username: HuaweiSoftX3000
       Session ID: 65
       Session Version: 65
       Owner Network Type: IN
       Owner Address Type: IP4
       Owner Address: 69.18.206.8
   Session Name (s): Sip Call
   Time Description, active time (t): 0 0
       Session Start Time: 0
       Session Start Time: 0
   Media Description, name and address (m): audio 0 RTP/AVP 0 8
       Media Type: audio
       Media Port: 0
       Media Proto: RTP/AVP
       Media Format: 0
       Media Format: 8
   Media Attribute (a): inactive

Frame 4 (520 bytes on wire, 520 bytes captured)
   Arrival Time: Sep 15, 2004 23:41:48.864245000
   Time delta from previous packet: 6.960477000 seconds
   Time relative to first packet: 7.908797000 seconds
   Frame Number: 4
   Packet Length: 520 bytes
   Capture Length: 520 bytes
Ethernet II, Src: 00:0c:85:97:5b:80, Dst: 00:0b:db:94:1e:e4
   Destination: 00:0b:db:94:1e:e4 (00:0b:db:94:1e:e4)
   Source: 00:0c:85:97:5b:80 (00:0c:85:97:5b:80)
   Type: IP (0x0800)
Internet Protocol, Src Addr: 69.18.206.8 (69.18.206.8), Dst Addr: 216.128.80.41(216.128.80.41)
   Version: 4
   Header length: 20 bytes
   Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
       0000 00.. = Differentiated Services Codepoint: Default (0x00)
       .... ..0. = ECN-Capable Transport (ECT): 0
       .... ...0 = ECN-CE: 0
   Total Length: 506
   Identification: 0x63c4
   Flags: 0x00
       .0.. = Don't fragment: Not set
       ..0. = More fragments: Not set
   Fragment offset: 0
   Time to live: 239
   Protocol: UDP (0x11)
   Header checksum: 0x2a6a (correct)
   Source: 69.18.206.8 (69.18.206.8)
   Destination: 216.128.80.41 (216.128.80.41)
User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
   Source port: 5061 (5061)
   Destination port: 5060 (5060)
   Length: 486
   Checksum: 0x7cdd (correct)
Session Initiation Protocol
   Status line: SIP/2.0 200 OK
   Message Header
       From: "3472201228" <sip:3472201228@216.128.80.41>;tag=as1b5cd2e7
       To: <sip:19177230306@69.18.206.8>;tag=b7c53fe6
       CSeq: 102 INVITE
       Call-ID: 789697e678776aab545459c817d8aaf3@216.128.80.41
       Via: SIP/2.0/UDP 216.128.80.41:5060;branch=z9hG4bK7dea1a59
       Contact: <sip:19177230306@69.18.206.8:5061;user=phone>
       Content-Length: 103
       Content-Type: application/sdp
Session Description Protocol
   Session Description Protocol Version (v): 0
   Owner/Creator, Session Id (o): HuaweiSoftX3000 65 66 IN IP4 69.18.206.8
       Owner Username: HuaweiSoftX3000
       Session ID: 65
       Session Version: 66
       Owner Network Type: IN
       Owner Address Type: IP4
       Owner Address: 69.18.206.8
   Session Name (s): Sip Call
   Time Description, active time (t): 0 0
       Session Start Time: 0
       Session Start Time: 0
   Media Description, name and address (m): audio 0 RTP/AVP 0 8
       Media Type: audio
       Media Port: 0
       Media Proto: RTP/AVP
       Media Format: 0
       Media Format: 8
   Media Attribute (a): inactive


clearly, the Hauwei device is not responding with valid media ports..  IMO, this is not an asterisk bug....
By: twisted (twisted) 2004-09-15 23:34:45

Closing since the problem obviously lies with the Huawei device's response with the media port at 0.  If this is not the case, please contact one of the bug marshals to re-open.

Thanks!  (and don't worry, i'm not gonna de-karma you, since i can confirm that it wasn't reporting the response in the asterisk console)