Re [Serusers] RTPProxy only fails for Public to Private communication
Vivienne Curran
vivcurran at yahoo.co.uk
Fri Apr 8 11:29:36 CEST 2005
Hello,
After further testing I have determined what exactly seems to be happening. When the private client initiates a call to the public phone and the public phone then rings back the private phone, everything is fine. However if after a period of time, the public phone rings the private phone, there is no audio.
Im presuming this is somthing got to do with the fact that the RTPProxy doesnt know where the audio should be delivered or something got to do with ports?? My registration messages should last for many days. I have included the 200OK message sequences below when 2092 (public) rings 2093(private) and theres no audio. I would like if someone could clarify if the "nortpproxy:yes" is appropriate and if the "c" field of the sdp is correct. Should the c field contain the public address of the natted client or the address of the rtpproxy??
BR,
Vivienne
84.203.148.14:5060 -> 84.203.148.146:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 84.203.148.146;branch=z9hG4bKe725.70eccca1
.0..Via: SIP/2.0/UDP 157.190.74.151;rport=5060;branch=z9hG4bK16f933418fead0
a2..From: "2092" <sip:2092 at 84.203.148.146>;tag=50ca3e7d33baf708..To: <sip:2
093 at 84.203.148.146>;tag=e7e4d9d22be835d6..Call-ID: 2dfa5ad2915bcf13 at 157.190
.74.151..CSeq: 55703 INVITE..User-Agent: Grandstream BT100 1.0.5.18..Contac
t: <sip:2093 at 172.16.3.31>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTION
S,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported: replaces..Conte
nt-Length: 152....v=0..o=2093 8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN I
P4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP 0..a=sendrecv..a=rtpmap:0 PCMU/
8000/3..a=ptime:20..
U 84.203.148.146:5060 -> 157.190.74.151:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 157.190.74.151;rport=5060;branch=z9hG4bK16
f933418fead0a2..From: "2092" <sip:2092 at 84.203.148.146>;tag=50ca3e7d33baf708
..To: <sip:2093 at 84.203.148.146>;tag=e7e4d9d22be835d6..Call-ID: 2dfa5ad2915b
cf13 at 157.190.74.151..CSeq: 55703 INVITE..User-Agent: Grandstream BT100 1.0.
5.18..Contact: <sip:2093 at 84.203.148.14:5060>..Allow: INVITE,ACK,CANCEL,BYE,
NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Support
ed: replaces..Content-Length: 174....v=0..o=2093 8000 0 IN IP4 172.16.3.31.
.s=SIP Call..c=IN IP4 84.203.148.146..t=0 0..m=audio 35014 RTP/AVP 0..a=sen
drecv..a=rtpmap:0 PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..
Vivienne Curran <vivcurran at yahoo.co.uk> wrote:
Another email - Sorry for posting so many timea! - Im sorry it was incorrect to says the messages are still the same. Since the extra fix_nated_contact hass been added in, the 200 ok message sent from SER to the public phone (i.e. the phone who is initiating the call) is different.
Below are the two 200 OK messages BEFORE the change was made:
U 84.203.148.14:5060 -> 84.203.148.146:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 84.203.148.146;branch=z9hG4bKf51e.b169be72
.0..Via: SIP/2.0/UDP 157.190.74.151;rport=5060;branch=z9hG4bKcd17ddd1b59ead
49..From: "2092" <sip:2092 at 84.203.148.146>;tag=aedc22bd5a3b510c..To: <sip:2
093 at 84.203.148.146>;tag=acd725e00242a605..Call-ID: 8ffc2d18b21870b3 at 157.190
.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100 1.0.5.18..Contac
t: <sip:2093 at 172.16.3.31>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTION
S,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported: replaces..Conte
nt-Length: 152....v=0..o=2093 8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN I
P4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP 0..a=sendrecv..a=rtpmap:0 PCMU/
8000/3..a=ptime:20..
U 84.203.148.146:5060 -> 157.190.74.151:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 157.190.74.151;rport=5060;branch=z9hG4bKcd
17ddd1b59ead49..From: "2092" <sip:2092 at 84.203.148.146>;tag=aedc22bd5a3b510c
..To: <sip:2093 at 84.203.148.146>;tag=acd725e00242a605..Call-ID: 8ffc2d18b218
70b3 at 157.190.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100 1.0.
5.18..Contact: <sip:2093 at 172.16.3.31>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,
REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported: rep
laces..Content-Length: 174....v=0..o=2093 8000 0 IN IP4 172.16.3.31..s=SIP
Call..c=IN IP4 84.203.148.146..t=0 0..m=audio 35016 RTP/AVP 0..a=sendrecv..
a=rtpmap:0 PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..
Now the messages are as follows:
U 84.203.148.14:5060 -> 84.203.148.146:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 84.203.148.146;branch=z9hG4bKb9eb.e87f3706
.0..Via: SIP/2.0/UDP 157.190.74.151;rport=5060;branch=z9hG4bK14d3c4f08f811a
a6..From: "2092" <sip:2092 at 84.203.148.146>;tag=12bc28408336e179..To: <sip:2
093 at 84.203.148.146>;tag=34216f9b980666a6..Call-ID: a29762e6f24c12e9 at 157.190
.74.151..CSeq: 21848 INVITE..User-Agent: Grandstream BT100 1.0.5.18..Contac
t: <sip:2093 at 172.16.3.31>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTION
S,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported: replaces..Conte
nt-Length: 152....v=0..o=2093 8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN I
P4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP 0..a=sendrecv..a=rtpmap:0 PCMU/
8000/3..a=ptime:20..
U 84.203.148.146:5060 -> 157.190.74.151:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 157.190.74.151;rport=5060;branch=z9hG4bK14
d3c4f08f811aa6..From: "2092" <sip:2092 at 84.203.148.146>;tag=12bc28408336e179
..To: <sip:2093 at 84.203.148.146>;tag=34216f9b980666a6..Call-ID: a29762e6f24c
12e9 at 157.190.74.151..CSeq: 21848 INVITE..User-Agent: Grandstream BT100 1.0.
5.18..Contact: <sip:2093 at 84.203.148.14:5060>..Allow: INVITE,ACK,CANCEL,BYE,
NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Support
ed: replaces..Content-Length: 174....v=0..o=2093 8000 0 IN IP4 172.16.3.31.
.s=SIP Call..c=IN IP4 84.203.148.146..t=0 0..m=audio 35002 RTP/AVP 0..a=sen
drecv..a=rtpmap:0 PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..
So the contact header is re-written when SER forwards the 200 OK to the calling phone. However like I said, when the public phone rings the private phone, the public phone can hear voice but the private phone cant.
Apologies for sending so many emails in a row.
BR
Vivienne.
Vivienne Curran <vivcurran at yahoo.co.uk> wrote:
Sorry I also should have mentioned that now when the public phone rings the private phone, the public client can hear voice but the private client can still hear nothing.
BR
Viv
Vivienne Curran <vivcurran at yahoo.co.uk> wrote:
Hi Greger (Sorry forgot to copy the serusers mailing list the 1st time),
I made that change and my on_reply_route now looks like this:
onreply_route[1] {
if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {
fix_nated_contact();
if (!search("^Content-Length:\ 0")) {
force_rtp_proxy();
};
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
Unfortunately when I restarted rtpproxy, SER and reregistered my phones, I still had the same message dump.
U 84.203.148.146:5060 -> 157.190.74.151:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 157.190.74.151;rport=5060;branch=z9hG4bKcd
17ddd1b59ead49..From: "2092" <sip:2092 at 84.203.148.146>;tag=aedc22bd5a3b510c
..To: <sip:2093 at 84.203.148.146>;tag=acd725e00242a605..Call-ID: 8ffc2d18b218
70b3 at 157.190.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100 1.0.
5.18..Contact: <sip:2093 at 172.16.3.31>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,
REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported: rep
laces..Content-Length: 174....v=0..o=2093 8000 0 IN IP4 172.16.3.31..s=SIP
Call..c=IN IP4 84.203.148.146..t=0 0..m=audio 35016 RTP/AVP 0..a=sendrecv..
a=rtpmap:0 PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..
Any more thoughts?
BR
Viv
"Greger V. Teigre" <greger at teigre.com> wrote:
Look at this:
U 84.203.148.146:5060 -> 157.190.74.151:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 157.190.74.151;rport=5060;branch=z9hG4bKcd
17ddd1b59ead49..From: "2092" <sip:2092 at 84.203.148.146>;tag=aedc22bd5a3b510c
..To: <sip:2093 at 84.203.148.146>;tag=acd725e00242a605..Call-ID: 8ffc2d18b218
70b3 at 157.190.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100 1.0.
5.18..Contact: <sip:2093 at 172.16.3.31>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,
REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported: rep
laces..Content-Length: 174....v=0..o=2093 8000 0 IN IP4 172.16.3.31..s=SIP
Call..c=IN IP4 84.203.148.146..t=0 0..m=audio 35016 RTP/AVP 0..a=sendrecv..
a=rtpmap:0 PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..
I assume this is where you get an error message. You haven't called fix_nated_contact() for this message, and in fact I believe there may be an error in the ONsip.org example where a line has been lost.
192. onreply_route[1] {
193.
194. if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {
195. if (!search("^Content-Length:\ 0")) {
196. force_rtp_proxy();
197. };
198. } else if (nat_uac_test("1")) {
199. fix_nated_contact();
200. };
201. }
fix_nated_contact() should go in between line 194 and 195.
Could you please confirm that this works? I will look at the config file.
g-(
---- Original Message ----
From: Vivienne Curran
To: serusers at lists.iptel.org ; greger at teigre.com
Sent: Wednesday, April 06, 2005 04:12 PM
Subject: Re: RTPProxy fails only for Private to Public communication
> Just as an extra : I have a sniff of the message for when a public
> client (2092)rings a private client (2093)included at the bottom of
> this email. I cant see anything wrong with them but maybe it will
> shed more light on the matter.
>
> Vivienne Curran <vivcurran at yahoo.co.uk> wrote:
> I changed the line modparam("nathelper", "rtpproxy_sock",
> "/var/run/rtpproxy.sock") to modparam("nathelper", "rtpproxy_sock",
> "udp:localhost:22222") and started the rtpproxy as ./rtpproxy -s udp
> from the relevant directory and this resulted in a series of
> "rtpp_command: no response from rtpproxy" and rtpproxy temporarily
> disabled" errors. If I return to the original mod param and start it
> as ./rtpproxy then it works but like I said when the private client
> rings the public client, I get "ERROR: send_rtpp_command: cant read
> reply from a RTP Proxy".
>
> Any further ideas? Has anyone on the mailing list experienced this? I
> am using the script given in the onsip getting started doc for 0.9.0.
> but am using ser 0.8.14.
>
> BR,
> Vivienne
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