[cisco-voip] RTP only one way?
Gerd Feiner
g.feiner at cablesurf.de
Sat Apr 30 04:48:59 EDT 2005
Hi Bryan,
thanks for your reply. But: there is NO NAT involved here, the config
for NAT stuff just sits therer in case there would be a box sitting
behind a NAT. To understand my topology:
The AS5350 sits on one leg of a uBR10012 while the ATA sits behind a
cablemodem on a cable-interface of the uBR - all non-private, fully
functioning, official, shining and polished public addressing :P There
is no single private IP involved, except the ones between the CMTS and
the cablemodem, but thats for management only. All CPE and the AS have
public IP-addresses - and this is what puzzles me the most.
Brgs,
Gerd
Am 29.04.2005 um 19:30 schrieb Bryan Deaver:
> 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
>> third-party authority to import, export, distribute or use encryption.
>> Importers, exporters, distributors and users are responsible for
>> compliance with U.S. and local country laws. By using this product you
>> agree to comply with applicable laws and regulations. If you are
>> unable
>> to comply with U.S. and local laws, return this product immediately.
>>
>> A summary of U.S. laws governing Cisco cryptographic products may be
>> found at:
>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>
>> 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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