[cisco-voip] RTP only one way?

Fri Apr 29 12:57:45 EDT 2005

OK, here we go.

#sh ver
Cisco Internetwork Operating System Software
IOS (tm) 5350 Software (C5350-IK9SU2-M), Version 12.3(13), RELEASE 
SOFTWARE (fc2)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2005 by cisco Systems, Inc.
Compiled Thu 10-Feb-05 04:17 by ssearch
Image text-base: 0x60008AFC, data-base: 0x61880000

ROM: System Bootstrap, Version 12.2(1r)1, RELEASE SOFTWARE (fc1)
BOOTLDR: 5350 Software (C5350-BOOT-M), Version 12.2(2)XB2, EARLY 
DEPLOYMENT RELEASE SOFTWARE (fc1)

mgw_muc_1 uptime is 1 day, 6 hours, 39 minutes
System returned to ROM by reload at 00:10:16 GMT Sat Jan 1 2000
System restarted at 12:14:08 GMT Thu Apr 28 2005
System image file is "flash:c5350-ik9su2-mz.123-13.bin"

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cisco AS5350 (R7K) processor (revision T) with 262144K/131072K bytes of 
memory.
Processor board ID JAE08512F9T
R7000 CPU at 250MHz, Implementation 39, Rev 1.0, 256KB L2, 2048KB L3 
Cache
Last reset from IOS reload
Channelized E1, Version 1.0.
Bridging software.
X.25 software, Version 3.0.0.
SuperLAT software (copyright 1990 by Meridian Technology Corp).
Primary Rate ISDN software, Version 1.1.
Manufacture Cookie Info:
  EEPROM Type 0x0001, EEPROM Version 0x01, Board ID 0x32,
  Board Hardware Version 3.34, Item Number 800-5171-02,
  Board Revision C0, Serial Number JAE08512F9T,
  PLD/ISP Version 2.2,  Manufacture Date 14-Dec-2004.
Processor 0x14, MAC Address 0x012048F32E
Backplane HW Revision 1.0, Flash Type 5V
2 FastEthernet/IEEE 802.3 interface(s)
130 Serial network interface(s)
120 terminal line(s)
4 Channelized E1/PRI port(s)
512K bytes of non-volatile configuration memory.
65536K bytes of processor board System flash (Read/Write)
16384K bytes of processor board Boot flash (Read/Write)

Configuration register is 0x2102

all the other information is attached in txt-files.

Brgds,
Gerd

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Am 29.04.2005 um 18:36 schrieb Bryan Deaver:

> OK, please get some debugs and show commands to better understand what 
> is
> happening.
>
> For a _single_ call, gather the below debugs.  Make sure that you have
> console logging disabled first and always nice to have timestamps set 
> to
> msec.
>     - debug ccsip message
>     - debug ccsip events
>     - debug isdn q931
>     - debug voip ccapi inout
>
> Also, when the call is connected and in this one-way audio state, 
> please get
> the output of 'show call active voice brief'.
>
> If you can please send the show version as well.  The complete 
> configuration
> would be good as well but you can send that privately if you want; with
> passwords removed.
>
> Bryan
>
>
>
>> From: Gerd Feiner <g.feiner at cablesurf.de>
>> Date: Fri, 29 Apr 2005 17:59:45 +0200
>> To: Kevin Thorngren <kevint at cisco.com>
>> Cc: "<cisco-voip at puck.nether.net> <cisco-voip at puck.nether.net>"
>> <cisco-voip at puck.nether.net>
>> Subject: Re: [cisco-voip] RTP only one way?
>>
>> Hi Kevin,
>>
>> no there aren't any routing issues.  The AS can ping the sipura and
>> there are indeed RTP-packets from the AS to the sipura - about 1 every
>> two seconds, while there are many many packtes from the sipura to the
>> AS5350 ... when debugging SIP the AS5350 also tells about opening a
>> recv-only audio-stream:
>>
>> Apr 29 12:41:56 x.x.x.x 42940: Apr 29 10:48:30.444: sipSPIAddStream:
>> AddStream in idle state to open a 'recvonly' media session
>>
>> this is why I digged into and found that rtp send-receive command and
>> its exactly what happens:  voip can speak and is heard on pstn, but 
>> not
>> vice versa.
>>
>> any ideas?
>>
>> Brgds
>> Gerd
>>
>> Am 29.04.2005 um 15:47 schrieb Kevin Thorngren:
>>
>>> Hi Gerd,
>>>
>>> Typically one way voice issues are due to IP Routing issues.  Can the
>>> 5300 ping the SIP Phone?
>>>
>>> This URL should help you diagnose the problem.
>>> http://www.cisco.com/en/US/tech/tk652/tk698/
>>> technologies_tech_note09186a008009484b.shtml
>>>
>>> Kevin
>>> On Apr 29, 2005, at 6:49 AM, Gerd Feiner wrote:
>>>
>>>> Hi there,
>>>>
>>>> we have an AS5350 and using as a SIP-Gateway to the PSTN.  Now, 
>>>> there
>>>> is an intriguing issue:  SIP -> PSTN voice is audible, and there is
>>>> an RTP stream from the SIP-device - SIPURA - to the mediagateway.
>>>> But there is no stream in the SIP direction coming from the AS5350.
>>>> I already found the
>>>>
>>>> voice rtp send-receive
>>>>
>>>> command - but it didn't do the trick. As of now, I wasn't able to
>>>> ascertain the source of the problem.  It doesn't matter who is
>>>> initiating the call, it's always the same effect.
>>>>
>>>> Don't know which part of our config you need, but here are a few:
>>>>
>>>> voice rtp send-recv
>>>> !
>>>> voice service pots
>>>>  fax protocol pass-through g711alaw
>>>> !
>>>> voice service voip
>>>>  signaling forward rawmsg
>>>>  fax protocol pass-through g711alaw
>>>>  sip
>>>>   rel1xx disable
>>>>   no call service stop
>>>> !
>>>> ...
>>>> !
>>>> interface Serial3/0:15
>>>>  no ip address
>>>>  isdn switch-type primary-net5
>>>>  isdn incoming-voice modem
>>>>  isdn sending-complete
>>>>  no cdp enable
>>>> !
>>>> voice-port 3/0:D
>>>>  bearer-cap Speech
>>>> !
>>>> dial-peer voice 1 pots
>>>>  tone ringback alert-no-PI
>>>>  application session
>>>>  incoming called-number 143677..
>>>>  destination-pattern .
>>>>  translate-outgoing calling 20
>>>>  translate-outgoing called 20
>>>>  supplementary-service pass-through
>>>>  no digit-strip
>>>>  direct-inward-dial
>>>>  port 3/0:D
>>>> !
>>>> dial-peer voice 2 voip
>>>>  tone ringback alert-no-PI
>>>>  application session
>>>>  incoming called-number .
>>>>  destination-pattern 143677..
>>>>  voice-class codec 10
>>>>  session protocol sipv2
>>>>  session target ipv4:x.x.x.x
>>>>  supplementary-service pass-through
>>>> !
>>>> !
>>>> dial-peer search type voice data
>>>> sip-ua
>>>>  nat symmetric check-media-src
>>>>  sip-server ipv4:x.x.x.x
>>>>
>>>> this isn't by far complete, but it seems to be the important part as
>>>> I figured.  In addition, I don't really understand all of the
>>>> commands set, most of it was from an example, part is from the 
>>>> ?-help
>>>> system and another part is from cisco's voice config guide ...
>>>>
>>>> Glad if someone could help.
>>>>
>>>> Brgds,
>>>> Gerd Feiner
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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