CCME calls to SIP ISP problem

Unanswered Question

I have a connection to ISP by SIP. My phone 7940 SCCP registereg on my 2811 CCME 4.1 So, I configured sip-ua authetication, assign ISP-number to my phone, configure registation server, BUT - I don't know how to configure sip proxy. My ISP requred to use sip proxy to all calls. When I call to outside trought sip, I see that replyes rtp packets go not from proxy, so - I have one-way voice issue. Problem that I can'nt see an other host - just proxy. So, I need get rtp-packets from proxy only.

My config:

!

dial-peer voice 6 voip

destination-pattern 8T

session protocol sipv2

session target sip-server

dtmf-relay h245-alphanumeric

codec g711alaw

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none

!

!

sip-ua

authentication username xxx password xxx

registrar ipv4:192.168.128.41 expires 3600

sip-server ipv4:192.168.128.41

!

!

My sip-proxy is 192.168.128.41

How can I configure 2811 to call trought sip proxy?

I have this problem too.
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Paolo Bevilacqua Fri, 03/07/2008 - 03:47

Hi,

there is nat between cme and tisp ?

That is usually the cause of one-way voice.

sip-proxy just means sip-server. Different people use different names for the same things, you don't need to worry about that.

Paolo Bevilacqua Fri, 03/07/2008 - 04:04

Which way do you have one-way voice ?

How do you know the rtp packets are using the wrong addresses, and what are the wrong addresses ?

Ok, ip-addressing

interface f0/0: 10.10.250.1/30,

GW: 10.10.250.2

SIP-Proxy: 192.168.128.41/29

I can ping 192.168.41 - no problem with it.

I execute "debug voip rtp"

3

036005: Mar 7 17:20:14 EKAT: RTP(22736): fs tx s=10.10.250.1(16518), d=192.168.128.25(11094), pt=8, ts=85BBDB0, ssrc=837FA01

036006: Mar 7 17:20:14 EKAT: RTP(25821): fs rx s=192.168.128.25(11094), d=10.10.250.1(16518), pt=8, ts=BEBEBFA8, ssrc=19DB75C3

So, you can see that my ip 10.10.250.1, outside - 192.168.128.25, BUT - I don't have a route for that host. My ISP said that access to SIP-proxy is enought for propper work. If I call - I can hear other side, but other side don't hear me.

Paolo Bevilacqua Fri, 03/07/2008 - 04:32

Default route should be enough for you to get to 192.168.128.25, or add one as necessary.

Of course you need reachability to yuor TSP's voice gateway beside SIP server, remind them.

Paolo Bevilacqua Fri, 03/07/2008 - 05:51

Not necessarily. If the TSP want, they install a so called rtp-proxy, also know as media termination point (MTP) in cisco parlance. The sip messages will contain this device address for media.

This is usually inefficient as it creates a bottleneck, on the other hand it can be desirable to get transcoding, address hiding, etc.

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