Hi all,
We have successfully managed to get Ser to act as a proxy between our backend SIP service and clients out on the Internet using the nathelper modules and RTP proxying. However, one of the features of our SIP service allows you to dial a number via a web interface. This appears to work by sending an INVITE from the SIP service out to the IP phone using the 'Diversion' and follow-me directives. The problem we appear to be having is that the RTP information is setup not in the INVITE but rather in an OK status message returned when we pick up the phone to dial out. From what I understand Ser does not look inside Status messages but only in the initial INVITE, hence when we use the web interface to dial out the SDP location is not being altered to point at our Ser proxy and therefore the RTP is not being proxied as we want.
We have tried various things but the issue appears to be that Ser will not look inside OK replies to INVITE's. We might be misunderstanding things here??? By the way all SIP messages are going through the proxy as desired... The only issue we have is with the RTP traffic.
Can anybody help us understand how to get around this problem (if in fact it is possible to get around it using Ser)
Cheers,
Steve
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generally, you either need a 3pcc b2bua or you base your service on REFER. The ctd script in examples supports the latter, as for example seen in serweb.
-jiri
At 04:50 AM 11/24/2003, Stephen Miles wrote:
Hi all,
We have successfully managed to get Ser to act as a proxy between our backend SIP service and clients out on the Internet using the nathelper modules and RTP proxying. However, one of the features of our SIP service allows you to dial a number via a web interface. This appears to work by sending an INVITE from the SIP service out to the IP phone using the 'Diversion' and follow-me directives. The problem we appear to be having is that the RTP information is setup not in the INVITE but rather in an OK status message returned when we pick up the phone to dial out. From what I understand Ser does not look inside Status messages but only in the initial INVITE, hence when we use the web interface to dial out the SDP location is not being altered to point at our Ser proxy and therefore the RTP is not being proxied as we want.
We have tried various things but the issue appears to be that Ser will not look inside OK replies to INVITE's. We might be misunderstanding things here??? By the way all SIP messages are going through the proxy as desired... The only issue we have is with the RTP traffic.
Can anybody help us understand how to get around this problem (if in fact it is possible to get around it using Ser)
Cheers,
Steve
This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments.
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Jiri Kuthan http://iptel.org/~jiri/
Check out:
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/nathelper.cfg?r...
look for the part beginning with onreply_route[1], this is the part that rewrites 200 OK reponses.
Ser is instructed to use the onreply part by using t_on_reply("1") in route[1] section.
Jan.
On 24-11 16:50, Stephen Miles wrote:
Hi all,
We have successfully managed to get Ser to act as a proxy between our backend SIP service and clients out on the Internet using the nathelper modules and RTP proxying. However, one of the features of our SIP service allows you to dial a number via a web interface. This appears to work by sending an INVITE from the SIP service out to the IP phone using the 'Diversion' and follow-me directives. The problem we appear to be having is that the RTP information is setup not in the INVITE but rather in an OK status message returned when we pick up the phone to dial out. From what I understand Ser does not look inside Status messages but only in the initial INVITE, hence when we use the web interface to dial out the SDP location is not being altered to point at our Ser proxy and therefore the RTP is not being proxied as we want.
We have tried various things but the issue appears to be that Ser will not look inside OK replies to INVITE's. We might be misunderstanding things here??? By the way all SIP messages are going through the proxy as desired... The only issue we have is with the RTP traffic.
Can anybody help us understand how to get around this problem (if in fact it is possible to get around it using Ser)
Cheers,
Steve
This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments.
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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Jan Janak -
Jiri Kuthan -
Stephen Miles