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.
dial-peer voice 6 voip
session protocol sipv2
session target sip-server
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
authentication username xxx password xxx
registrar ipv4:192.168.128.41 expires 3600
My sip-proxy is 192.168.128.41
How can I configure 2811 to call trought sip proxy?
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.
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 ?
interface f0/0: 10.10.250.1/30,
I can ping 192.168.41 - no problem with it.
I execute "debug voip rtp"
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.
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.
So, In sip-communication, RTP going between endpoints - clients, right? Like in h323-communication, and redirection servers or proxy servers doesn't influens to that?
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.
Ok, now I know about sip more than I know in this morning =)
Yet another question: what diffirence between proxy and redirect servers in IOS configuration? In what case sip work with redirect servers and in what - with proxy?