[cisco-voip] Cisco 2600 VIC-2BRI - no sound

Wed Mar 3 04:49:47 EST 2004

          Guan,

 as the voice actually travels like a RTP stream, definitely one will see no
RTP packets if no voice travels between the
 parties. Possible problem for this situation are:

1. codec mismatch
2. firewall (RTP is actually UDP, make sure on the XP you have closed all
firewalls).

Better on the XP run a protocol analyzer like Ethereal (www.ethereal.com)
and you will see what is actually going on.
Ethereal is fully capable of "reading" VoIP sessions like H.323, SIP and
etc.

----- Original Message ----- 
From: "Guan Yang" <guan at unicast.org>
To: <cisco-voip at puck.nether.net>
Sent: Tuesday, March 02, 2004 12:47 AM
Subject: [cisco-voip] Cisco 2600 VIC-2BRI - no sound

> This is one of those "why is there no sound?" questions. My setup is a
> Cisco 2611 with NM-2V and VIC-2BRI-S/T-TE. I'm on a Euro-ISDN
> (basic-net3) connection. The config's below (phone numbers changed). I've
> also attached output of various diagnostic commands below, as well as some
> comments at the bottom.
>
> isdn switch-type basic-net3
> !
> !
> !
> voice class codec 1
>  codec preference 1 g711alaw
>  codec preference 2 g729br8
>  codec preference 3 g729r8
>  codec preference 4 g723r63
>  codec preference 5 g723r53
>  codec preference 6 g711ulaw
> !
> voice class codec 2
>  codec preference 1 gsmfr
>  codec preference 2 gsmefr
> !
> interface BRI1/0
>  no ip address
>  no ip route-cache
>  isdn switch-type basic-net3
>  isdn incoming-voice voice
>  isdn answer1 32580600
> !
> voice-port 1/0/0
>  compand-type a-law
>  cptone GB
> !
> dial-peer voice 2 pots
>  application session
>  incoming called-number 32580600
>  destination-pattern T
>  direct-inward-dial
>  port 1/0/0
> !
> dial-peer voice 1 voip
>  application session
>  destination-pattern 32580668
>  voice-class codec 1
>  session protocol sipv2
>  session target sip-server
>  session transport udp
> !
> sip-ua
>  sip-server ipv4:192.168.1.106
> !
>
> 192.168.1.106 is my laptop with Windows XP and a softphone. I've checked
> that the softphone (both SJPhone and eStara) work fine by dialing
> another Windows box with a softphone. Procedure:
>
> 1. I dial my mobile phone by entering sip:12345678 at 192.168.1.102
> (12345678 represents my mobile phone number, and 192.168.1.102 is the
> router's IP address).
>
> 2. The call goes through fine. However there's no sound, whether I
> speak into the mobile phone or my laptop's microphone. Output of "show
> isdn status":
>
> ISDN BRI1/0 interface
>         dsl 1, interface ISDN Switchtype = basic-net3
>     Layer 1 Status:
>         ACTIVE
>     Layer 2 Status:
>         TEI = 105, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
>     Layer 3 Status:
>         1 Active Layer 3 Call(s)
>         CCB:callid=16, sapi=0, ces=1, B-chan=1, calltype=VOICE
>     Active dsl 1 CCBs = 1
>     The Free Channel Mask:  0x80000002
>
> show voice call:
>
> 1/0/0 1
>       vtsp level 0 state = S_CONNECT
> callid 0x8008 B01 state S_TSP_CONNECT clld 12345678 cllg
> 1/0/0 2  -         -  -
> 1/0/1 1  -         -  -
> 1/0/1 2  -         -  -
>
> show voice dsp:
>
> DSP  DSP             DSPWARE CURR  BOOT                         PAK
TX/RX
> TYPE NUM CH CODEC    VERSION STATE STATE   RST AI VOICEPORT TS ABORT  PACK
> COUNT
> ==== === == ======== ======= ===== ======= === == ========= == =====
> ============
> C542 001 01 gsmfr     4.1.38 busy  idle      0  0 1/0/0.1   NA     0
> 13049/20747
>
> show voice trace 1/0/0:
>
> 1/0/0 State Transitions: timestamp (state, event) -> (state, event) ...
> 8420.532 (S_SETUP_INDICATED, E_CC_PROCEEDING) ->
> 8420.904 (S_PROCEEDING, E_CC_ALERT) ->
> 8422.014 (S_ALERTING, E_CC_CONNECT) ->
>
> show rtp call:
>
> No Active calls found
>
> show call active voice brief:
>
> Total call-legs: 2
> 11FC : 842050hs.1 +151 pid:2 Answer 12345678 active
>  dur 00:02:42 tx:7209/237897 rx:4408/145464
>  Tele 1/0/0 (28) [1/0/0] tx:162700/44080/0ms gsmfr noise:-71 acom:45
> i/0:-67/-76 dBm
>
> 11FC : 842053hs.1 +145 pid:1 Originate 32580600 active
>  dur 00:02:42 tx:4408/145464 rx:7209/237897
>  IP 192.168.1.106:16384 rtt:0ms pl:10380/17600ms lost:903/1392/1219
> delay:70/70/210ms gsmfr
>
>
> Note that I have one voice-class with G.7xx codecs and another with GSM
> codecs. I've tried both. I've also tried SJPhone on Linux.
>
> Could the problem be related to the fact that there's no RTP
> connection going on? What's weird is, if I dial digits on my softphone
> (I guess DTMF or RFC2833), I can actually here the DTMF tones on my
> mobile phone, but not the other way around. I guess this is because
> the DTMF is transferred over RFC2833 and outputted by the Cisco box.
>
> Another thing that's weird is the "i/0 -67/-76 dBm" -- what are normal
> values for these two figures?
>
> I hope that someone can help.
>
> Guan
>
> PS. I've previously posted this message at comp.dcom.sys.cisco with
> slightly different output.
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