| Summary: | ASTERISK-05511: Multi-Linking and UDP/SIP | ||
| Reporter: | Shane Spencer (whardier) | Labels: | |
| Date Opened: | 2005-11-08 15:21:58.000-0600 | Date Closed: | 2011-06-07 14:03:15 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | This could go for IAX as well, however since thats primarily used for trunking it would not be the right area to discuss this. I see how many VoIP servers may have a muli-link setup for redundancy or partitioning bandwidth from multiple providers, for sip calls from dynamic peers outbound it would be required to know what link, rather what source IP to use in that situation. Recording the source IP (the local destination for the registration) for dynamic sip devices and setting the source IP for calls to those privileged SIP devices would help distinguish which source IP, and in turn which gateway, to use for SIP packets and the RTP stream from then on. I may be mistaken that this already works plainly.. or that if the proxy/registrar on the sip device is set up correctly it will negotiate the path correctly when a call is placed from the server to the device. Any information reguarding this would be helpfull. S | ||
| Comments: | By: Olle Johansson (oej) 2005-11-15 09:45:51.000-0600 Please take this discussion to the mailing list, it's meant for discussions whereas the bug tracker is not :-) /o | ||