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C3825 as SIP-to-SIP Gateway doesn't work.

Ellad Yatsko
Level 1
Level 1

I tried to setup Cisco 3825 as SIP-toSIP GW between customer's Voice Gateway (Quintum AXF, 80.251.131.27) and "RTU" Soft-Switch (cluster's IPs: 80.251.131.132 and 80.251.131.133 and internal ones 10.0.99.121, 10.0.99.221 and 10.0.99.99 - shared Heartbeat IP).

When I connect Quintum directly to RTU (making corresponding settings on RTU and Quintum) - all works fine! The matter is not in Quintum or in RTU. This is verfied.

Cisco3825 (80.251.131.28) accepts SIP INVITEs and sends ones on the same interface (one interface configuration what is "highly recommended"). But it doesn't help. Call looks like in the following trace got on RTU (10.0.99.111 is my phone for simplification):

17:44:50.930821  10.0.99.111 -> 10.0.99.99   SIP/SDP Request: INVITE sip:4996540908@10.0.99.99;user=phone, with session description

17:44:50.933458   10.0.99.99 -> 10.0.99.111  SIP Status: 100 Trying

17:44:50.957366 80.251.131.132 -> 80.251.131.28 SIP/SDP Request: INVITE sip:0008014996540908@80.251.131.28;user=phone, with session description

17:44:50.976924 80.251.131.28 -> 80.251.131.132 SIP Status: 100 Trying

17:44:51.534246 80.251.131.28 -> 80.251.131.132 SIP/SDP Status: 183 Session Progress, with session description

17:44:51.535698   10.0.99.99 -> 10.0.99.111  SIP/SDP Status: 183 Progress, with session description

17:44:51.596882 80.251.131.28 -> 80.251.131.132 SIP Status: 503 Service Unavailable

17:44:51.597308 80.251.131.132 -> 80.251.131.28 SIP Request: ACK sip:0008014996540908@80.251.131.28;user=phone

17:44:51.598557   10.0.99.99 -> 10.0.99.111  SIP Status: 503 Service Unavailable

17:44:51.599091  10.0.99.111 -> 10.0.99.99   SIP Request: ACK sip:4996540908@10.0.99.99;user=phone

This trace has been got from RTU side. At the moment when 183 Session Progress is coming on the phone connected to Quintum's analogue port one ring is heard (single). When 503 Service Unavailable is coming I get short rings on my phone, Quintum's Phone become silent, because Quintum judging on its traces gets CANCEL from Cisco.

I have tried a handful of IOSes, with "T" of without it, but effect is always much the same.

How-to? Help, please!... :-)

Router#sh run

Building configuration...

Current configuration : 3109 bytes

!

! Last configuration change at 12:25:08 UTC Tue Jun 25 2013 by eyatsko

!

version 15.1

service timestamps debug datetime msec

service timestamps log datetime msec

service password-encryption

!

hostname Router

!

boot-start-marker

boot system flash:/c3825-adventerprisek9-mz.151-3.T2.bin

boot system flash:/c3825-adventerprisek9-mz.124-24.T2.bin

boot system flash:/c3825-adventerprisek9_ivs_li-mz.124-24.T2.bin

boot system flash:/c3825-advipservicesk9-mz.151-3.T2.bin

boot-end-marker

!

! card type command needed for slot/vwic-slot 0/0

!

no aaa new-model

!

!

dot11 syslog

ip source-route

!

ip cef

!

!

no ipv6 cef

!

multilink bundle-name authenticated

!

!

voice-card 0

!

!

voice service voip

ip address trusted list

  ipv4 80.251.131.128 255.255.255.224

allow-connections sip to sip

sip

!

!

voice class uri RTU1 sip

host 80\.251\.131\.132

!

voice class uri RTU2 sip

host 80\.251\.131\.133

!

voice class codec 1

codec preference 1 g711alaw

!

!

voice translation-rule 801

rule 1 /^000801\(.*\)$/ /\1/

!

voice translation-profile 801

translate called 801

!

crypto pki token default removal timeout 0

!

license udi pid CISCO3825 sn FHK0903F413

archive

log config

  hidekeys

username eyatsko privilege 15 password 7 0322531D001A354042030056

!

redundancy

!

interface GigabitEthernet0/0

no ip address

duplex auto

speed auto

media-type rj45

!

interface GigabitEthernet0/0.16

encapsulation dot1Q 16

ip address 80.251.131.28 255.255.255.240

!

interface GigabitEthernet0/0.18

encapsulation dot1Q 18

ip address 10.0.99.34 255.255.255.0

shutdown

!

interface GigabitEthernet0/1

no ip address

shutdown

duplex auto

speed auto

media-type rj45

!

ip forward-protocol nd

!

!

no ip http server

no ip http secure-server

ip route 0.0.0.0 0.0.0.0 80.251.131.17

!

logging esm config

!

!

control-plane

!

mgcp profile default

!

dial-peer voice 801 voip

translation-profile outgoing 801

huntstop

destination-pattern 000801.+

session protocol sipv2

session target ipv4:80.251.131.27

dtmf-relay rtp-nte

codec transparent

fax-relay ecm disable

fax rate 9600

fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 2 fallback none

no vad

!

dial-peer voice 901 voip

modem passthrough nse codec g711alaw

session protocol sipv2

incoming called-number .+

incoming uri from RTU1

dtmf-relay rtp-nte

codec transparent

fax-relay ecm disable

fax rate 9600

fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 2 fallback none

no vad

!

dial-peer voice 902 voip

shutdown

modem passthrough nse codec g711alaw

session protocol sipv2

incoming called-number .+

incoming uri from RTU2

dtmf-relay rtp-nte

codec transparent

fax-relay ecm disable

fax rate 9600

fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 2 fallback none

no vad

!

!

line con 0

exec-timeout 1440 0

logging synchronous

login local

line aux 0

line vty 0 4

exec-timeout 1440 0

logging synchronous

login local

transport input all

line vty 5 15

exec-timeout 1440 0

logging synchronous

login local

transport input all

!

scheduler allocate 20000 1000

end

Router#

Kind regards,

Ellad Yatsko

21 Replies 21

This is exactly my thought..However looking at the traces it is not obvious..The only thing that I can see is that the Quintum endpoint does not include the fmtp (format parameter) for the digits it will support using rtp-nte for DTMF transport

Received:

SIP/2.0 183 Session Progress

Call-ID: A858991F-DD7A11E2-810A93BC-5853A7A8@80.251.131.28

Content-Length: 227

Content-Type: application/sdp

CSeq: 101 INVITE

From: <4996540000>;tag=4DD3A88-242B

Require: 100rel

RSeq: 7000

To: <4996540908>;tag=50fb831b-ba

User-Agent: Quintum/1.0.0 SN/0030E1103326

Via: SIP/2.0/UDP 80.251.131.28:5060;branch=z9hG4bK13D2F

Quintum: 070e0000000c00000006001e03808081

v=0

o=Quintum 103 7050 IN IP4 80.251.131.27

s=VoipCall

c=IN IP4 80.251.131.27

t=0 0

m=audio 10358 RTP/AVP 8 101

c=IN IP4 80.251.131.27

a=rtpmap:8 pcma/8000/1

a=ptime:20

a=rtpmap:101 telephone-event/8000/1------------no dtmf events advertised here

a=sendrecv

NB: According to rfc2833..This is not  mandatory:

    
   Receivers MAY indicate which named events they can handle, for
   example, by using the Session Description Protocol (RFC 2327 [7]).
   The payload formats use the following fmtp format to list the event
   values that they can receive:

   a=fmtp: 

   The list of values consists of comma-separated elements, which can be
   either a single decimal number or two decimal numbers separated by a
   hyphen (dash), where the second number is larger than the first. No
   whitespace is allowed between numbers or hyphens. The list does not
   have to be sorted.

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Message was edited by: Ayodeji oladipo Okanlawon

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I don't think it would fail the call over the DTMF transport.  Not specifying the digits is technically supported by the RFC.  CUCM/CUBE used to have issues when the other side didn't specifically advertise digits but this has been fixed in any recent code.  Also, the call would never actually fail in those cases.  Just DTMF would fail.

I think it's probably the other rtpmap line for the actual call codec of G.711alaw.  That would definitely cause the call to fail if the CUBE didn't understand it.  I found a similar case that was resolved by upgrading the CUBE but they didn't attach a bug ID and I'm not finding anything obvious that added this support.

Looking at RFC 3551 and the IANA specification for RTP/AVP for audio codecs, I observed that the encoding is specifiec in CAPS.

8 = PCMA...

I also observed that CUBE sticks to this as well as CUCM and other sip devices I have seen. However this ip endpoint is using lowercase.....That cound be the issue? Is that what you were referring to?

a=rtpmap:8 pcma/8000/1

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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I'm referring to the encoding paramter which is the 1 in this example:

a=rtpmap:8 pcma/8000/1

This is pretty uncommon but definitely doesn't violate RFC4566

a=rtpmap: / [/]

For audio streams,  indicates the number
         of audio channels.  This parameter is OPTIONAL and may be
         omitted if the number of channels is one, provided that no
         additional parameters are needed.

I've actually never seen this set before.  Usually, the encoding parameter is just left out since it's pretty much always only setting up 1 audio channel.  It looks like some versions of 15.x CUBE code don't like this being set.

Hello, guys!

:-)

"voice-class sip rel1xx disable" helped! Inspite of "no CUCM".

Kind regards,

Ellad

Out of curiosity, can you send a sample of a working log..debug ccsip messages

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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Sure! :-)

IPPhone -> RTU -> C3825 -> Quintum -> Phone

4996540000 -> 4996540908