We have cisco IP telephony infrastructure with 2 call managers and around 1000 phones (call managers in head office and phones are distributed in 4 branch office). We recently started facing an issue with voip calls on one of the existing branch office where in voip call from head office becomes one-way after some times of call establishment sometimes to some users (intermittent). During this issue calls will be still running untill users hangs up, however it will be one -way. Phones are deployed to this site 2 year ago (mainly 6941 series phones with latest firmware) started with 40 phones and now expanded to 120 phones in this site. Issue happens only with head office to this site calls. Internal site calls, PSTN calls from this site works perfectly. Since only this site is having this issue, bit confused on making any changes in call manager configuration. Head office to this site uses G.729 codec to communicate. All phones registering to head office. Please note that firewall (Juniper SRX) has been recently replaced with the old Juniper SSG20 in this site. Cisco 4507E is the current core switch at this site.
Your advice will helpfull in tracking and fixing this issue.
Please check you have opened all RTP ports between all phones subnets in both sites
You can start openening full ip. but again confirm you are opening for all ip phones subnets
This is routing issue between the sites. Check both site routers. Ping each other and use static/dynamic routing
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Nisaar, when you say:
"where in voip call from head office becomes one-way after some times of call establishment sometimes to some user"
does that mean the call actually starts of as a fully functioning normal call with "two way"voice?
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Thank you for the quick reply.
Yes, call starts normally with two way voice for sometime and suddenly becomes one - way. Issue happens some -time to different users. When we drop the call and call again, it will continue to be a one - way voice untill we restart the remote branch user phone then calls works fine.This proves that its not a routing issue
We are trying to collect the wireshark streams of both sides, since issue is intermittent its became bit hard to capture.
Please advice if any other way of stream capture to trace the call drop error code.
May we know what side/phone cannot hear audio sound? If the issue recurs, kindly browse the IP phone to check the send/receive packets.
I think PSTN and Internal extension calls (within site) work fine because it does not goes through Juniper SSG20 (correct me if i am wrong) , only the calls to head office having this one way audio (intermittent) problem , because there is something wrong on this firewall which is passing signaling messages but blocking the RTP stream of one side at that moment from a source ip and clears it once you reboot the phone.
I will check on firewall first whenever user complain oneway audio and collect ip address of both end ip phone and see if there is any thing related with these IPs.