Understanding SIP/RTP call flow

WRKNonCCNP

I have a Cisco 3825 running Call Manager Express, and two SIP phones registering to it. The registration goes fine and i can call between the phones as i would expect, but when i look at the RTP stream using wireshark, the RTP stream does not go between the IP phones directly, but through the CME router.

So my questions are these:
1. It was my understanding that typically, the RTP streams would go directly between the endpoint phones. How does SIP tell the endpoints where to send their RTP streams under these conditions?

2. How do i tell the CME router to tell the phones to send the RTP to each other, rather than through the CME router?

Any help with this would be appreciated.

shodown

Are the phones in the same subnet? if not they would have to go through the CME for routing?

You are correct the RTP steam is just suppose to flow between the devices. If you have a copy of a show run we could take a look.

WRKNonCCNP

Sorry for the slow response. The phones are not on the same subnet, but the CME router is not part of the path between the endpoints. Routing between the phones directly works.

Here is the config:

voice service pots
!
voice service voip
allow-connections sip to sip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
registrar server expires max 1200 min 300
!
voice register global
mode cme
source-address 10.99.10.10 port 5060
max-dn 20
max-pool 10
load 7960-7940 P0S3-8-12-00
authenticate register
tftp-path flash:
create profile sync 5700285131555181
!
voice register dn 1
number 9034
name Phone1
label 2142101001
!
voice register dn 2
number 5715
name Phone2
label 2142101002
!
voice register pool 1
busy-trigger-per-button 1
id mac 001D.A266.DACC
type 7960
number 1 dn 1
dtmf-relay sip-notify
username user2 password cisco
codec g711ulaw
no vad
!
voice register pool 2
busy-trigger-per-button 1
id mac 000A.B7B1.3638
type 7940
number 1 dn 2
dtmf-relay sip-notify
username user3 password cisco
codec g711ulaw
no vad
!
!
ip tftp source-interface GigabitEthernet0/0
!
interface GigabitEthernet0/0
ip address 10.1.1.250 255.255.255.0
duplex auto
speed auto
media-type rj45
!
interface GigabitEthernet0/1
ip address 10.99.10.10 255.255.255.0
duplex full
speed 100
media-type rj45
!
tftp-server flash:SIP41.8-4-2S.loads
tftp-server flash:dsp41.8-4-1-23.sbn
tftp-server flash:jar41sip.8-4-1-23.sbn
tftp-server flash:cvm41sip.8-4-1-23.sbn
tftp-server flash:cnu41.8-4-1-23.sbn
tftp-server flash:apps41.8-4-1-23.sbn
tftp-server flash:term41.default.loads
tftp-server flash:term61.default.loads
tftp-server flash:P0S3-8-12-00.loads
tftp-server flash:P0S3-8-12-00.sb2
tftp-server flash:P003-8-12-00.bin
tftp-server flash:P003-8-12-00.sbn
!
control-plane
!
dial-peer voice 1002 voip
destination-pattern ..........
no voice-class sip localhost
voice-class sip outbound-proxy ipv4:172.30.201.17:5070
session protocol sipv2
session target ipv4:172.30.201.17:5070
session transport udp
incoming called-number ..........
codec g711ulaw
!
!
sip-ua
no remote-party-id
set sip-status 404 pstn-cause 17

shodown

so if they are on different networks where does the routing take place at?

WRKNonCCNP

Routing is done by routers in between. Let's use the following diagram as reference:

CME Router.....[CME]
........................|
........................|
......................[R1]
....................../...\
..................../.......\
.................[R2]......[R3]
...................|..........|
IP Phones...[SIP].....[SIP]

I know it is crude, sorry. All of the subnets between routers can talk to one another. Routing appears to work correctly and i can ping between the subnets the IP phones are on. I just need to know how to tell the phones not to use the CME as a proxy, or tell the CME to not act as a proxy. Thanks.

Unforg1ven

i always thought RTP stream would go through a media relay instead of direct 1:1

hermeszdata

Two questions...

What IOS are you running on the CME router?
What features do you have active? Gateway/Gatekeeper, MGCP ...

Those answers may very well answer your question.

WRKNonCCNP

I'm using 12.4(24)T3 Adv. IP Services. The gateway command is not enabled. I don't see the gatekeeper command as an available option. To my knowledge, no MGCP is running either. From what i can tell, just SIP.

shodown

can you post the wireshark capture?

WRKNonCCNP

My CME router is 10.99.10.10, SIP phone 1 is 10.1.22.100, SIP phone 2 is 10.1.102.100. This capture is after a call has been originated from 10.1.22.100 and the CME router is sending the INVITE to 10.1.102.100, you can just see the communication between the CME and SIP phone 2. The file is a pcap renamed as a doc.

WRKNonCCNP

It seems to me that the phones themselves would have the required DSP resources, but that may not be the case, as i am not that familiar with voip and exactly how the IP phone interacts with the CME router.

So, is the SDP part of the INVITE message what is used to tell the endpoint where the other end of the RTP stream should go? That is my understanding. If that is the case, how does the CME router handle that? Shouldn't the CME router tell the originating phone to use the IP of the receiving SIP phone, rather than the IP of the CME router, for the RTP stream?

shodown

I took my DSP comment off as I"m not to sure at this point.

[FONT=&quot]debug voip ccapi inout[/FONT]
[FONT=&quot]debug ccsip messages [/FONT]



Those should help you narrow down the problem.

WRKNonCCNP

Thanks, i'll give those a shot.