[cisco-voip] RTP only one way?

Bryan Deaver bdeaver at cisco.com
Fri Apr 29 13:30:06 EDT 2005 

Wow, you really scrubbed the config and debug output.

>From the show call active voice brief, it appears packets are being sent
out, though at a rate smaller than the inbound rate.  It appears that you
don't have vad enabled on the sipura device where you do on the 5400.  Add
'no vad' on dial-peer 2 and you should see fairly symmetrical output in the
packet count.

Your sip-ua command 'nat symmetric check-media-src' appears to better
indicate what your typology looks like.  Someone already mentioned that
one-way audio is common in firewall/nat typologies.  The usual symptom is
that the audio works in the direction in->out but not out->in which I am
guessing is going on in your configuration.

Since we cannot see the addressing you are using, I recommend making sure
that the address you see in the show and debug output match what you expect
to happen.  I also suggest that you get an ethereal trace before and after
the NAT device and I expect that you will find the problem there.

If you are still having issues, please open up a TAC case so that we can
look more closely at the details of your setup in a less public forum.

Bryan





> From: Gerd Feiner <g.feiner at cablesurf.de>
> Date: Fri, 29 Apr 2005 18:57:45 +0200
> To: Bryan Deaver <bdeaver 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?
> 
> 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"
> 
> 
> This product contains cryptographic features and is subject to United
> States and local country laws governing import, export, transfer and
> use. Delivery of Cisco cryptographic products does not imply
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> 
> If you require further assistance please contact us by sending email to
> export at cisco.com.
> 
> 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
> 
> 
> 
> 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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