06-25-2013 06:56 AM - edited 03-16-2019 06:04 PM
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
Solved! Go to Solution.
06-26-2013 01:51 PM
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=4DD3A88-242B4996540000>
Require: 100rel
RSeq: 7000
To: <4996540908>;tag=50fb831b-ba4996540908>
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
06-26-2013 02:22 PM
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.
06-26-2013 02:46 PM
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"
06-26-2013 05:18 PM
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.
06-26-2013 11:21 PM
Hello, guys!
:-)
"voice-class sip rel1xx disable" helped! Inspite of "no CUCM".
Kind regards,
Ellad
06-27-2013 12:22 AM
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"
06-27-2013 12:28 AM
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